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  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.5.0-5" date="1370692800">- Make GeoIP.dat -&gt; GeoIP-initial.dat symlink in %install, not %post,
  and don't %ghost it
- Run geoipupdate silently in %post and cron job
- Create empty database files for %ghost to work with old rpm versions
- Don't try to use noarch subpackages on old rpm versions
- Update %description to mention database updates
- Drop outdated README.Fedora</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.5.0-6" date="1370865600">- Update sub-package requires main package for geoipupdate script</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-7" date="1384344000">- Don't download the database in %post (#1029807)
- Don't download GeoLiteASNum as it's no longer provided.</changelog>
  <changelog author="Daniel Mach &lt;dmach@redhat.com&gt; - 1.5.0-8" date="1388145600">- Mass rebuild 2013-12-27</changelog>
  <changelog author="Daniel Mach &lt;dmach@redhat.com&gt; - 1.5.0-9" date="1390564800">- Mass rebuild 2014-01-24</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-10" date="1467201600">- Added base IPv6 database to GeoIP package
- Added GeoIP-data subpackage with all other databases (#1201857)</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-11" date="1467201601">- Fixed rpmlint errors</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-12" date="1529409600">- Removed clean section
- Resolves: #1589860 - Update GeoIP legacy databases
- Resolves: #1409060 -  Missing symlink at the GeoIP-update package</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-13" date="1533038400">- Related: #1409060 -  Missing symlink at the GeoIP-update package</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-14" date="1550491200">- Resolves: #1666767 - Remove geoipupdate binary from GeoIP</changelog>
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  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.5.0-5" date="1370692800">- Make GeoIP.dat -&gt; GeoIP-initial.dat symlink in %install, not %post,
  and don't %ghost it
- Run geoipupdate silently in %post and cron job
- Create empty database files for %ghost to work with old rpm versions
- Don't try to use noarch subpackages on old rpm versions
- Update %description to mention database updates
- Drop outdated README.Fedora</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.5.0-6" date="1370865600">- Update sub-package requires main package for geoipupdate script</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-7" date="1384344000">- Don't download the database in %post (#1029807)
- Don't download GeoLiteASNum as it's no longer provided.</changelog>
  <changelog author="Daniel Mach &lt;dmach@redhat.com&gt; - 1.5.0-8" date="1388145600">- Mass rebuild 2013-12-27</changelog>
  <changelog author="Daniel Mach &lt;dmach@redhat.com&gt; - 1.5.0-9" date="1390564800">- Mass rebuild 2014-01-24</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-10" date="1467201600">- Added base IPv6 database to GeoIP package
- Added GeoIP-data subpackage with all other databases (#1201857)</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-11" date="1467201601">- Fixed rpmlint errors</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-12" date="1529409600">- Removed clean section
- Resolves: #1589860 - Update GeoIP legacy databases
- Resolves: #1409060 -  Missing symlink at the GeoIP-update package</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-13" date="1533038400">- Related: #1409060 -  Missing symlink at the GeoIP-update package</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-14" date="1550491200">- Resolves: #1666767 - Remove geoipupdate binary from GeoIP</changelog>
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  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.5.0-5" date="1370692800">- Make GeoIP.dat -&gt; GeoIP-initial.dat symlink in %install, not %post,
  and don't %ghost it
- Run geoipupdate silently in %post and cron job
- Create empty database files for %ghost to work with old rpm versions
- Don't try to use noarch subpackages on old rpm versions
- Update %description to mention database updates
- Drop outdated README.Fedora</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.5.0-6" date="1370865600">- Update sub-package requires main package for geoipupdate script</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-7" date="1384344000">- Don't download the database in %post (#1029807)
- Don't download GeoLiteASNum as it's no longer provided.</changelog>
  <changelog author="Daniel Mach &lt;dmach@redhat.com&gt; - 1.5.0-8" date="1388145600">- Mass rebuild 2013-12-27</changelog>
  <changelog author="Daniel Mach &lt;dmach@redhat.com&gt; - 1.5.0-9" date="1390564800">- Mass rebuild 2014-01-24</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-10" date="1467201600">- Added base IPv6 database to GeoIP package
- Added GeoIP-data subpackage with all other databases (#1201857)</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-11" date="1467201601">- Fixed rpmlint errors</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-12" date="1529409600">- Removed clean section
- Resolves: #1589860 - Update GeoIP legacy databases
- Resolves: #1409060 -  Missing symlink at the GeoIP-update package</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-13" date="1533038400">- Related: #1409060 -  Missing symlink at the GeoIP-update package</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-14" date="1550491200">- Resolves: #1666767 - Remove geoipupdate binary from GeoIP</changelog>
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  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.5.0-5" date="1370692800">- Make GeoIP.dat -&gt; GeoIP-initial.dat symlink in %install, not %post,
  and don't %ghost it
- Run geoipupdate silently in %post and cron job
- Create empty database files for %ghost to work with old rpm versions
- Don't try to use noarch subpackages on old rpm versions
- Update %description to mention database updates
- Drop outdated README.Fedora</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1.5.0-6" date="1370865600">- Update sub-package requires main package for geoipupdate script</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-7" date="1384344000">- Don't download the database in %post (#1029807)
- Don't download GeoLiteASNum as it's no longer provided.</changelog>
  <changelog author="Daniel Mach &lt;dmach@redhat.com&gt; - 1.5.0-8" date="1388145600">- Mass rebuild 2013-12-27</changelog>
  <changelog author="Daniel Mach &lt;dmach@redhat.com&gt; - 1.5.0-9" date="1390564800">- Mass rebuild 2014-01-24</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-10" date="1467201600">- Added base IPv6 database to GeoIP package
- Added GeoIP-data subpackage with all other databases (#1201857)</changelog>
  <changelog author="Jan Safranek &lt;jsafrane@redhat.com&gt; - 1.5.0-11" date="1467201601">- Fixed rpmlint errors</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-12" date="1529409600">- Removed clean section
- Resolves: #1589860 - Update GeoIP legacy databases
- Resolves: #1409060 -  Missing symlink at the GeoIP-update package</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-13" date="1533038400">- Related: #1409060 -  Missing symlink at the GeoIP-update package</changelog>
  <changelog author="Michal Ruprich &lt;mruprich@redhat.com&gt; - 1.5.0-14" date="1550491200">- Resolves: #1666767 - Remove geoipupdate binary from GeoIP</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-15" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-16" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-17" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-18" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.2.12-19" date="1518177600">- Escape macros in %changelog</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-20" date="1531396800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-21" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-22" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 1.2.12-23" date="1571313600">- Remove rpath from showimage binary to remove rpmbuild warning</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-14" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-15" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-16" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-17" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-18" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.2.12-19" date="1518177600">- Escape macros in %changelog</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-20" date="1531396800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-21" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-22" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 1.2.12-23" date="1571313600">- Remove rpath from showimage binary to remove rpmbuild warning</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-14" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-15" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-16" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-17" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-18" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.2.12-19" date="1518177600">- Escape macros in %changelog</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-20" date="1531396800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-21" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-22" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 1.2.12-23" date="1571313600">- Remove rpath from showimage binary to remove rpmbuild warning</changelog>
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  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-16" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
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  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1.2.12-19" date="1518177600">- Escape macros in %changelog</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-20" date="1531396800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.12-21" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
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  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 1.2.12-23" date="1571313600">- Remove rpath from showimage binary to remove rpmbuild warning</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
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</package>
<package pkgid="6d057c2ef35a75186ad40e0fbfdca110017fbd6405463f9e7d80688bff47f14d" name="amiws-debugsource" arch="x86_64">
  <version epoch="0" ver="2.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1673697600">- Updated to release 2.3.1</changelog>
</package>
<package pkgid="99722ba0b2bc0dd13bf55e5ed3e74b00e8e3f796fb71e39f080263056db84b6b" name="amiws-debugsource" arch="x86_64">
  <version epoch="0" ver="2.3.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1673697600">- Updated to release 2.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.2-1" date="1678363200">- Updated to release 2.3.2</changelog>
</package>
<package pkgid="db4d1fc7fd720b4f68e449afc882b8607304a9176567040c6067830d84cca19f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.10.0" rel="1.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a0f6c5f267ec88c4a2abce6fbd04d387ee63d181f142195f4e858da5e83c7e86" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.11.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="24f5c54b9fbfe1d3d07934280b3cedb7a308424e98a706b4445661c15c318061" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.11.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f55440e5ffd16aa0a4090a1f8a8de2dbc1490f2725152d0f709ef27da9fe8e83" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.12.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="54747e3b7fea776863d09cbae4b10dd8befeecc729217fb4b3851ca7dda0c0bd" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.13.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="40dd64289ccb33bed3f5f2a43cc02647d45857982d12b9ac205d9f660cee9216" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="956f66b74010afe1d911365a3106169f3e1b82cf8e44db612a94fe5b32c8a41d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.14.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d2a59680b0e15fc280f1ae88e00c90d3620984b09e4652c87d8489f9251fff96" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6eec9ead147ea73c34f093fc2b830bd453ffde01a6001273e323a313711f9058" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="440bf2811f846f6bba79f2a5825aaa610ac9b4b1d1a9c7e0c981afb76c15d7bf" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4a43e9b8421d37af41e3cb891866657b0affc99d69c0cb902f78447d575e4906" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.16.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e9d6386d8e2a102b320c64cee3702f5dee0334970a9b7391bb701f9e51847281" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.16.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fe2fbf1b5a768b594169db65184ba44ff75efcbc19378dec5274142243bab268" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0ec22d174c8e0a3c9f641aedbfc51d85e99a709224a30e54e4fb2b9b667816eb" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="38dc470a25a50fe4f057942c4601a630d6ee3a100a55188f532ab6fd45453062" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5a5e98bc78d0608e854f923e2fccfa17cd4e37122da32d7ea5633b2f3c11ec7f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.19.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3566e38e330e978f5f853a3ac30d215f6b0a49fc8a9660e3ef515c3d887ecd6f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="aa29fc0463832132fac329d49c2242898de6ae7473f1ca07a84420ca7021212c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.20.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="d9818a85c699fdc00b3006a964d468ebf95b1b70b7ac23ce0a5c8816b35f3a34" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.21.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="aef16b43520c896e6b3462460a7438ec00cbea56ba869f3adb7adfd92ed8f95b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.21.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="28e6bde4b2658bcc74058a93f6454e6fbd0cefbad4675f541007ce080e7b23e2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.22.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1b637f4901025d9ba86c1fa2b5b30534721edd2748abb1264d9738b3b791f4ea" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.23.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
</package>
<package pkgid="9b1eb4a26af62194550058bc2ed5f90c4c43e7f2d33142865533bff62dcebc77" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.24.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a413bd9a48d25d71d53c73146dc25c03b830b4c4d7e9579081b3c907706599a0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.24.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="826a834e54067d94c5125cdeb5956119a2337b3a5e4b328de1344db2ac5ed29e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.25.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="70a7eb04fc0e790ea7091f6e1467d2a2390c44ea2a6d171a1e73cf7296fded03" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.25.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f5028e2fadf9394bda13bdd38d0418d394a81ef20e02935c46f33ad11ba375a0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.25.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e08dea89d99dc53aeefcea0f481da21323d3014cca27535fbcae02314d89e927" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.25.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cde4336624fdf7911ff7614540b1ef2a426ecc3f17d65b905fad5bb0a8f8a875" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.26.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a47ebb8a302bea079d3ea37c40af8197a31d4c2beffa54073dc21144108ca30b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.26.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b4bda35c9b9dc75bdba05fe9c55336c34d5cb4857a5e4928b757f3f74dbd4cff" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.27.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9c77ea7b7dd1032df352c92ea3340886d3d3b094e8f0b2f1d258b30e63ef6a49" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8abc7aea03050f3b95a7719fd21cc9e2931ea50a434195924e1238a068ce8ad7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="284adb459fd6d9fa758369468d7301e91aa55bc274544d7e12bb49ce8777d748" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="e3438447ace06cabfc62c62df1d786b835bfb959b595fa9cb3d670b6e8808a03" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.29.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6516e0e9735c4e081d28b7bf06875e4fbe5fbad83dd0dd7eb27c53c6a03096fc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.29.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="10f5fc33b0fb948f4e3c18830fbc63e7cbf5032017ac9d281acbf4697da97a86" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.30.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option &lt;E2&gt;&lt;80&gt;&lt;9C&gt;peer_supported&lt;E2&gt;&lt;80&gt;&lt;9D&gt;
-                    (Reported by Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn&lt;E2&gt;&lt;80&gt;&lt;99&gt;t be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: &lt;E2&gt;&lt;80&gt;&lt;A6&gt;may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6e91f753d29005737b381932ddb50e1f6f8cd1ea640572e5ccef91fa390e27be" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.30.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option &lt;E2&gt;&lt;80&gt;&lt;9C&gt;peer_supported&lt;E2&gt;&lt;80&gt;&lt;9D&gt;
-                    (Reported by Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn&lt;E2&gt;&lt;80&gt;&lt;99&gt;t be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: &lt;E2&gt;&lt;80&gt;&lt;A6&gt;may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 16.30.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/16.30.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 16.30.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.30.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/16.30.0...16.30.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16.30.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list
-              before applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="f4f6d00aeced046500f98ca712a03eb0b25ff2f8a4fc8db200b03f0e97334989" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.6.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.0-1" date="1550232000">- Update to upstream 16.2.0 release for bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
</package>
<package pkgid="5ecfab712003843df6273092bd545486b1e6a8e1681588c3609dab0cc009451a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.7.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6b346cfb2b19ce02b11a49a9c098d1f784e65f349f9cc96bbea31cb55ad7df73" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.8.0" rel="1.el8"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2d804f3dcab1244d862ab05e3e9b15bdd9c0a7ac0e5e098a8270c2b01a2cc9c8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.9.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl) 
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes) 
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="07054a2807add00fc4b4997ccef0aa408c1b1ff4c9c99874da624736a3b02306" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.0.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="daa605d0ec897f2aa0c27210f9748b0488d008fa43ebffc220bdc605adf6536f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.0.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fe037c87658602a5deae260d2b77b925630014ee3add81e261cb14cd1700cfac" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.1.0" rel="1.el8"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0cd869c86d6fe56c1d029ab21d867faf81a051d66d7a0e3c67f95052d97df5b5" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.2.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="490b03999bc1cd2392d797f32884e954a9bdf2a0420632cdf515a4c3f1e6a123" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.3.0" rel="1.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8675f17915037016fb103c6cfaa3fdbf4f8cc547d74296231ae3481d3076c918" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.4.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7669c7b1854ee51371e7e238c3bcb7a8108f79caf43b43fd7e715c3a1969a1e6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.5.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="84fe6a9c8c6356d3847376e4525322c954f71727203b6096445b85f5e3b68033" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.5.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c819612e873f3f02536ffe641b6d9ef9b46ec6f952cc1ddba6c8a64730d6ab57" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.6.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="57a897df36e4179394370763a03903627b332fd7c97a4f377173ab76ce6edaab" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="102363cfc9cc8a45343db1e6566d8f3b84d567c6d4f1f0bb39463541fa48738e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cafa4a686e6c828ec3797e2e501f46c0401a4687cf00802be40822c07bf8ae96" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2dd526708dcb77a532d902a8fdec464dd8d30e1e7710ac0c0edc56b94d7a5330" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a18a9eac737d972e41c8f721c88ef28b7a4fd520a45aba5e77ce707d2d596781" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2806ec6f7dabdfc11b8a5863bf9c4b9dbb1f166c7af09a6937ac26f4c52cef67" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c088015cecc0d71d1f4e6c9537fbb9e92c07eeb82522e8a102c91e97d1454431" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="13e0e9221a30da871dd1b00e027d825c71fca1da05779ddf0a3d9998772da3cf" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.4" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1ef5b2f603110c07d074c33a98cf737bf0e055e36f1ebd1b9e5ffcc64f359389" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="74ee9775567aaef4cfe4278eb6fcd24b93dc1a0aa225443918a66e7ec7a168ea" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c3b3b5c10859f767a1fa4c490cf5545f0ea739529ec1cedece3c3b45481219b2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="146d87314c731209a99bd1504b2d681b0d737f226d88afa4ebb24de59654640a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.1.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1270f6ba6f527aaf5ffee9a8611e17bf9bcd47686581338caeffaafa0f957d19" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="87402101f1a9324a4e452cbaeed497eca07dd46a2e6db5ebd0fbf58dcacdb7b1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3533def5131ace4732e15fb099b627ad45c106c8582a553b05d952b547714516" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f392e8379978c6b465de3ecf5000617ae979d3ec8a4a9540853130ec59ec5d33" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.11.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d56d40929842a76062953443ffc1fb9f2b981fccf58f1d28c87cbd0aafef21f6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.11.1" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="028a18c008465ac444f966ab6524445d8e4b8deb82bfc3acfb3b0d6e15d6ec73" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.11.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3789b6fc6b9ccfdc285274abf8057b99c783809d9d7307bcdcf3b64e4809ad29" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.11.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ea828ccf0d0046f1c1971e6418400d4b0f67101d199d88da70f158b47ab4f976" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">* Thu May 12 2022 Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1
- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f00006eeceaf54fc824a0a9546795d5f52aff831c1bc90422498bc9d10107b89" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.12.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="996370e1503cc77ef8f4d50587f0304c8118e5a7f1292498840cfe69830f94f9" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.13.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e70545f02c1527b735ef6c707e7945a0caf1dc10bf20af796d97a4b08b73135c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6420165af1bd86d1188801967ddf765ff797fcda5c87fddbf8e3d29a317c09f5" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="2ef4e4f21a0d93cb43f035ea0bc7509ac3dcd63875b957861cfda9d28f73a7f4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="d93f7eb69c6e3a2058729ce56313123dc19b90f8c03ff6876e42168fd28d7b63" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="532ecfb8dc012e745a00aa2721b095e7f19d4b39a5626e12e3eef2e7f2f80c7c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cfcb5a9f1065dc4fdbcb08bf7ac08883cc3f5cbabeff581e2de2c5e28ffbadb8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="17514497c103c9dcdbf95ea3b8c8896a9b1cf6f3544a365a86fe586207665f9e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="73f954fc8044d2184f3a64bd3c14a0c617c71c89b5169ffe493b2ba34ebefa88" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.17.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="06120d2670eb009bcbe7e66961f8bd30977c29a30e8dfb23d3a01b4eb5fcfd96" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
</package>
<package pkgid="edf8a3d9e26b566a4d365a3852da610471851cac215c6960bf572c87bab2008a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.18.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="9ec1d3f27e6bdfd783d247f8868b3aac7403626b3346684498666c79fa439ba1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="a5098f674e452d7cff1ea42a44879839899fc79d3d8ecbba263229d4031429d4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a5ea018451dff4514fc735864b7d9f9f32df46bd77c2ba0bc2789a42aea381d3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6b4f0a7afa5a0c378735f3cee7f3a37d1c9fed525b4fab8912d18ccfbb4cf694" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.2.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ca30ee0f2840827389d87078621544ee40ae39a0fb8bbe774c9fc8fa9b4d8951" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="8c2431e36a1f4cd93f61f94093c6aa44c2575c363d593db525de402b5050b9fc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.20.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="766247dd909e1556cdee36a4569e52025eefb062dbd22c88ba7c9fe1be9c6d52" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.20.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="8cdaa8e9c6207eeed522181bfb891067442caba117b96a7614796997e7c72122" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.21.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="830d35eeab3e305d33538a843b66998d5a3cb69a1b739058ea51004984324b1b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.22.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="477a256825a832115f0266c5e61ee9018677ab3eb590f84423a0fb195df717af" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.23.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="04a383148eeb07afd12c3f38cdfeee47b3ade6ce97f973d038fe4b40bc4a20a5" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.23.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="d050ecac3432a5f693ab0b436c21f9b7f3d580408677f8f33f55d5b8ba59c760" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.24.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="0a4b311ae9b3d03585deb52cdc34a4c8c6bca75840e7d28b57d7e4d017f0db2d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.24.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="c5f29c7c4bca01961ba662d62c656b41ad33a4b5b0d6f99cd336d1f0eb0dd131" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.24.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="2b3b393af3a0f50fa5cc6e01520aba6df88da4caf43e64950504a28dc6d8682f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.24.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="5ee0afcd5c6b931b906113a21fbe7f40cdad12996615822a6537e84241478ded" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.25.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="83378f958e314be622ac6f737a8f8fffea4deebc67e6067501a186392d7eef0a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.26.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
</package>
<package pkgid="e24d51ec5c5c693fe6b5135c5b2f8515f68e6ddca25137cb80a4fcf962689e47" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.26.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="43d2fa647f8ac981a46d2f1ae9bc9c0685e6c5a6dc14b2e36fee8f795953ed91" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.26.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="4a7d94e45c7fd22d17cf86cbf830ea24fba44d038cd64d0cd3503ec0721ebb81" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.26.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.3-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="69ea1d012eb2314407a9ccdfcd0c537d89315d329133d9532c71019b4da5f4f4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.26.4" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.3-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.4-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.4.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.4 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.4
-
- Change Log for Release asterisk-18.26.4
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.4.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.3...18.26.4)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.4.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-557q-795j-wfx2](https://github.com/asterisk/asterisk/security/advisories/GHSA-557q-795j-wfx2):
-    Resource exhaustion (DoS) vulnerability: remotely exploitable leak of RTP
-    UDP ports and internal resources
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-557q-795j-wfx2: Resource exhaustion (DoS) vulnerability: remotely
-                       exploitable leak of RTP UDP ports and internal resources
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    pjproject: Update bundled to 2.15.1.
-
- Commit List:
- ------------
-
- pjproject: Update bundled to 2.15.1.
-
- Commit Details:
- ---------------
-
- pjproject: Update bundled to 2.15.1.
-            Author: George Joseph
-            Date:   2025-08-25
-
-            This resolves a security issue where RTP ports weren't being
-            released causing possible resource exhaustion issues.
-
-            Resolves: #GHSA-557q-795j-wfx2</changelog>
</package>
<package pkgid="92cb4ee299781e235a5f745cdc436ea637dc3c804a2d330fa0fd1c9f65dbe93a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="14e3ac5e5e6eb0ae5a912e7e76cc514f48d9035d2a7bc86383e55c6a36417e99" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="dd625addc242f82f14d603974403ed5f60bc693c46f8f5f0f62dbce1072f6ef8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="01238539046642b303a30a6050085c642b0d4526fca93c016453d5b173ccd1ad" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7fe4309ff032ad3a6c1efe85e11a9e5411e1954710a093ac36306cef0b14d27c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="cbf981dd786722b212f0c467b59cc929e9424f40a2c9f0b10f5064fc423b8d17" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.6.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="cb0df8cb74550ec23e92e5b1609852d2ac533fa00a2e6e0294fdf0624bb1c5d5" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a03329d8ced87253b7aec50e33468e8225626781b50e12e6ab99acdfc6484355" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.7.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ff40e6dcdb37e8547d3a170803a2e758fe2b773d2f398938334b3976a81567fc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c60f77f28b7407e551d5e4847c57d015bdd1c93cdcc30997da4702b695c27fb1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="72e6e03de91c808d24fb7b8e5ce4a070a541ec65cd393f14da5f20c2af854c43" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f0e679507a5e409b694a3d8f2dd072d22d23f2b97be85a916cf932966a16b4f2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a4979910c16128dbbb244fad2c67e88f91ae51047b6545c9d531ff817083ccd1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1fde4db260c3140c358dbd9be37dc63e4e2a80ef43b404c830c25cd040badcf0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e6d358b3ce36a0fef15e8336e8239e923423f2b1468b47e9351421bbde117f01" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7b39e6cb5b87397e50a04ac4c5a14b9dc22d8863957476ae1463fff7ff124f82" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="976029b95cfd101244bbe0cdab271ad84852a0eaba614ce5ca2887c6867ca3e4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="29eb4d97ad48e1bf54ed4ecb60f5759e2d70cd3637f12ce7e194aa9a459f1c2f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.3.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="05280c13451ca6c57780803a2f7d19163491bda8f76168d65d8566fce7cb0d82" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.3.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="546cbe5c9b20be14bf9ab0b8a5f3820bbe19624500f280144205980fd45bef8a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="62d7dae26eb200403626fc19a3f2d6356ce6a323161e5f775c70d57e4ff4a351" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="613cb18045babfed5d59c9680cfabf16ae74d6a9ded6dc5195df5e7b11af693a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="297cced01978a1d7c87b4c58fbb093dd05358868d79fb0599dca34ab012a908e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="98364c830a1c97bb0826da6fe6a08fb991025b671a727417478d920bce471684" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="85eb50ab44f788a4ff0e70d547cffed324da4d268030335142ff559d33aa8cf2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="8f9a213ae6ec4cb4222a35a6131014a4f95af05603106a04c43e6c9eb252a21c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="94807ee13a35a89657df9aa4c9af6db517df658a5feaa105234d770532725d74" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.7.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0d92a741932659a26a8e4447bb44172e9302ef4046f5b426a8c91fb6e5091a9b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896001">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8fa672d21f6a4ccda9dd95a8c28b19f18d28b2ac47b0d051b2d312d7328feb17" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896001">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 19.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/19.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 19.8.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/19.8.0...19.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-19.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="81eb2e57e7fef7d372babc67f1ffae667851ff0b8334abc892fdba170804e552" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d47d95285c572c189c8e1a310ce53724e5b79e8b0ba83102f0c4f7aa13cc7408" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="de4a7438b2f310ab6f4ed10bea5342270df293ba369e7675769f08f6df971b3e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2d7108a78d455f86b87fc53cd31d2145cf08a1eea0b116884b24d6ac204d6505" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="8e053f96f74301f974f977a95b5b1e6bb1fdf7a483d1a45a42c3426b47839683" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="dd5f4a6d36ef7517a0fecfef91b552d99de717bed5bf7e4645532cf75db5b2b6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="bfd5770038aa374d042081475848bbf4b8267d207d167f62bd2655abbbb6eca4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="2a37896dc806b730c84501ce8c445ca82bb48617d59782a94995a859b8ae3f62" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="3782d1a6c0e39bdcd7b1d5093b678858f1c2c91cedcac0c2b48c91d481811aa7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="4195edc0bde8c04769dee86027ee184d92dce257deb325fcc118c88320d657a1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="cfe7b702d2a00ce5ae1e8e82058b5a85c47c0950a79191ea1468375224871342" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="3719ee543d834f405b4821c8f7be34bb6ff69a548b0ebca135d839078552c2db" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="b00372c85c3680098701aded974d9bcb3c0174fc6f8c8f574b5f4bd9b63a5d04" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="a1b6652601c0838c99d5bf870a9a9987d2fb84888aceccf5cc50ab39ec768060" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="d0a0d7565403d63a006fd9052a5c27720ac3a4a958c33c396961c19739f014b4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="6db9b31fd0fd6bf862eef8b95ed5b64718d70d2316bfb3facd5d732708d88deb" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="1558cd9fd422a665748329a7bffef9c3e4bb0bc51c3e39140bcb5bb9c2cd02fc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="7214eb44577bd0ae67913c283bffae12050df6aa4d61b1cf72fb212cb24498dc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="b437b9740a05ff7dcb8854ac27ba4c955a827cbc02e867129842bd55a29f40b2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="1ac8954a176b716fec8e8899712e148ecc42215094fd5db807db457b7cef93f9" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d0448f9fd8a2e050fce07b489c273ceda31cf399de5b99a931083c888660f765" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d5f4ad483232c8642e8a3a25d93d954415a04b3a487b2b03e14ff4db20e0a02f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="eea4b15b215a1071b1510f57299c6f0bda578812aa7bde61fb282d955fbd39b2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="0393c7634e8c0b97048d3bfa67f239c20bbd4c1ca1cbd080d0201c44b1f120ca" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
</package>
<package pkgid="5770d8b14c2a195377866e04aa73eff96a356d21d7b5db96007d2c47a5980f88" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="05e53f20fe99529151cdf2c58ad58e471204990d02f8b9c3c3fbc8e9c084aed6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="d6370eb8f22f8f6ed0c086fbe106b58d0deba81fdd103b71d9cf68d5f6c089f9" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="71feaed5328eac0c60b5765748af8f61c034c75938d40109b50b76e93cec314d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="59bfa9b33d779c0bb4343cc1d4cc69f94bb92f62c0019bb0dee93914947b93ad" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="40ccf2d08621417d9589c007c9b95a53b605f2c36f7030f990c0953da943a152" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="d751b2cc48cc51e13a0308b45cf5bf39fff285f98e52def1f4faf6390e2f17c8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="697cded1d14ac0ed88b38b56dd1926c43ed0fa375e344c244fe7a033ea28d5ab" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="e424bb803a35da0c3e90282c6759dc15dd2417dbefa4c0e5ef140d6b8c0ad6e1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="07df067fe108d8f33c9cedb03ddbea3bb8f89e20d8d6f668dfde09ae35336b1d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="1f49d19c2aa6416c51abcdaf2f415260b1ea02b8978c826128a3d6b0bc6e5660" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="da89cca52972179bf0728dfa485aba5b6cdf832d197860e61c6423bafd6c3ab0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="6aeb929d024818ed711953e682686a385a23269fa1bc48d5fcfc3c51201cb558" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="d6de86f76233bd3d38715a082102c0beffc8a3956bafee981278abed743e61ba" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="e6466e09e2e41f2f684b8bb4f2cfa0f0a7d2d0bd485e23fb4f7595dd69eda26e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="6679e73fc0c3dba8100274652b1051e1278ee18af7ecc623acca5c4b62eab29d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="96a2a864f69d077ee4cdb4ce1e64fbb2c19ba3e94452fb27b539fa3898864b3e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="9b3277c251afe1fba8b5ec51bfacd41d0abb2a512c1591d496f4eaded7f16585" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1747915201">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="749ec0386abad5c55cbcd82df64bcaa5457b3e80733f445d5072b6e3ddd67bd7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="0b39b50c1aea44c6f595c85af76cde17c58b08fafb25c26efb58c0a352fa2a94" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="da1020a353ea5ba67f2524c257f525a221b4a236a098c328940ed0df13cf76b6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="4d4c22b3f760f4a7420c1e5f43e128746d78a2d0fe3b939ec01a4f54cc4835d1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="4f37bfc98dc2ed374c140388620ce5138571795ae896d1d527ce47c489578982" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="556daf73c0a023880ada4a7d23d63d83947b591bf97ee0edef85a1d307f3348a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="39013e4946044a615c679e8b6a02be2bf47b538ab4e4725d483b5e86aadbc14d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="6aca38c935e61fee83b2c6667feba1073706c8d4921f7ff532007a6ec8ee95bc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="b492cc27d51b356520134f2dade2b339883bf2bf476da6921c1a673eccd4283d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="e3c682707600595cf7a82bc3e07f6fd9fdccade2ada92e5fd155dcaa293b3d6c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="f7df55e2d8487d7edad732fbe02d1c3153cc0437a2651a7094cffbea7d271adb" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="6c25ac927f0071a66c9b2cb86fe6c6710d8fdb2aa09e4c5130db10160b775461" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="c19640cbf5104a70ce3a3fba24d988b06af0cf5c167974eae091c00a53f32af5" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="aeb8bbc8d08c035ea1884bb174007e5c3f058afc7125b93926bf4fe49a3098a1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="8d359eaa042d35293dcc21bd732f7eff8972f1fc64faf543abb6ff8134bbcd15" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="4bd0fd61b8209259e3f93fde30df2828a702eb6ac825a1b5bb68a549ca0169eb" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="811b79c7a8d6746d1566ba9c287d24d772de24edecb641168dd507b6f97f2ded" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="33222d67dfd0ba93dca5bc73492f249889e407fe5eeffdac5cb2dc012dce5be3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="92d681aac9e4b1eaa2d7a2be01af2741fa8207bf218273d646414c57e0ef70d2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="ff94e22c0503c70086a0664c448e0099ccae48e1d61217d4e5287b3c8a0138ec" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="f0102296b5339126eca6dd046035a61c0d7213fee4a62ff9340522e56fb34e5d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="88ff9808216bbc5c02b24228ffb5b4e74c936e3696c21ad5669d09f944c41f5e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="a09a06814b57a0e00069015ffdba44147cc9df3b34c65ab23b64696ffca06fb9" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="08f1eeefb83505fc58cd8093ac082897ecede8a51668f976820cf81cccbc7b98" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="c4c3efbf212ccff98e27a68412f099924b8fbfe193fb05917bb81cd8fa7c8e5d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="e6d31c25b15e1ddfbf0573890ca0091dbd768c89a0145235d7705f7a7e26a2a5" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="b46ae2a777b97c036d0856d530274a1834bdace946a3ecc451c547b1c628d204" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="4d213519a97417a409570ae06dab09d8a5dc8bbcec46f84e0bfb4d0d3f22be9f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="ccb4ea5bb3b5acdb5502ad38f9c287b62c3e1a160478d0bf1de23de46c0b3438" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="ded97d828d3afc19dff45accb848e8bfab559e76cb618470e27d0c6975aab157" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="b6d92c8376b312a9b4ab04d74d5f4ef754fef97df21ab617feaeebf769f7df9d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="50d0e1f2f2b376e4c826a66610fbb5e604955ce6db9c362c90ebeff0c13a0c5b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="1becc6ba2020e97a63c4481a96da1d7456656472397799ed7c955ebc534e8134" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.5.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="986a365276b32d4327f6d7787af2021c7340c7e8e8974f87f0b254ee76675960" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="6b68d17971bd2b3413bc28aca7363fe9d69a4d11bd633d101f00f40afa3a0e75" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="510adb49acd0ac0a3a5d510b9ea20291decd02a21662025cc2947aa86437ef11" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="09b6b57b81d284872b135d6a8df319661a8df667320174477eb2bcca5956c64d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="7dc18200f8e242972807e4eb4bbed2d58ad0f2b9e69506343e8c4d146834ab58" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.8.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="55f4733f044d7fcf8f298b6c0ab31caf152b5f17dd98023767ebe644258391a8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="7ee199c0465729ac656adbc55b60f3e0274f5435777cb2eb7216b4f5b25c8679" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="83a2600e713ac9e9263022290c5a863d9090cba6b4f878a5b40cd0d6f1387bb3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="6b31cc41161a7c5c99ba41e7d2a64f8c877996a4dcb6d1c67548a5b569a1b52b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="92a7799cf39e37d68b5781ed4ae997e49fdca775dcf8af13bd9aadb4bead8447" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="bdd7fb985653cbabe5ffaa32d17c41e6b46405a4fca70a5132f7c4b49ccdd1d7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.2.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="8f9344e0b585a5fff931d1768f57fe0c7253408181602211d700fe622b27098e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="f6e4c6ded4dab66ea9956585f7c6296f31e410e3875e1d5f04c58ed52753d819" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="f3a2ec7233468b052ade86905b72a9cbfd95eea7f8ffb202db92cf93718a9397" name="asterisk" arch="x86_64">
  <version epoch="0" ver="23.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="013abf97a76b72aa5498d9b73c992e21e6d3a4cb86cacbefa84c98f524a1672f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">* Thu May 12 2022 Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1
- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c5d681e28fa5c11e36157cb426c6f18c0cd1766a4dc81bedca75d991da64ae50" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.12.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4ec3a7bfee35ec720a06093e59c42e837d48db9e9e6cb9db294f41962ba35b86" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.13.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="00fd81f371d419f0f98842e788a9926b4d82f8a93933ba003053e9b84634d867" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a0ee005307278c4bcd4932378b70cec25bca4df0c24ab2d0a107b72f5b565fd2" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="a3fdcdcb3a9f1c5122973fa2d16985d1153f88d42e93e04a092e9b93a211f784" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="593cbb06c17703d66e62d8453f695a590d85962e2c67a7fa8d7cf8d51a535b4f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d860c46c9e0bd113f3a3dbf9cfbcd53f01aad135c91ddd6a6820a752e9c60769" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c34e323ee57ca4ec8c083c06de9c6d6caf2f388be58c3c5b590b2aab7549fda8" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="12ce64c93486e374c55f7a76c32aff4e7c4962c43a28c72b45e400d07a7935eb" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="55c8d5bb31f1b1840a544ed2ceb4b95e3578272ecdbd8fb9d4fb9572f395a447" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.17.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="49d1b14fa5430ed76170efdfb364d1d69b4ae3d285aebacdc91dfe0f3348e8c4" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
</package>
<package pkgid="61a037443605f1c9924fe5decf4ef8202e56ef2c2d01816feca0484a04d5e6ee" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.18.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="e659a11a43ed5777377b2723d3664177cd5b64f3b581332bff19e56a8cfaa8be" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="c9af6e8b95297b50922ef3d1f7015f0c68916344c8187b5a3cf46e3849f402c9" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="80499ffd55a4e3da20dcb92e386da90998b388de5059f899b2725c2db37ceb2c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.20.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="ff380bfae161914d5925243ee55194f9b3c3db35893d9f72d0652703e216a201" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.20.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="2672f39b1ccc70534933d922c777286f1d43e5f820bf6cb862cdffc5ac53dc8c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.21.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="97959f29107e769826aa157e526a38ae0de3162ec89b238e7512cf289c72c37b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.22.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="dfdca8cfd127fa21271ae19d2f51d87a4c29b90d6065f379497217951e137347" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.23.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="ade9b9eb4a6e007fa27bdbaf8ee751ac5aac90f9d6deef34948cd649b261d583" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.23.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="18207ed873e72f16d529311ec335e8aede2b99caaef8c61efb288a0f9929ac91" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.24.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="85005e0145708afe0527258f55380a611c2f32622215b5b66fdf33d30045fee2" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.24.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="42891efd2b883ea99d941ac99a4db1349ea8824bcab35d5cd6f5344c4d4966fe" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.24.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="1e515135bc87448cb01b2f4211502cc1c50c2e93ad2ace5a0be9d09b31051309" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.24.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="f90a41cb87ee4c44adbe425305bf6732351afe77b1f19686561009b18a23d5b6" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.25.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="8711dabca7ba58ace4b5eba73dc1ee37284e5a14f5e531881236dff353b39f0d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.26.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
</package>
<package pkgid="5842f9f65de1228b9ab89ed52002d51cf53698baa12ab6eab3cce9674698e73d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.26.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="8cf5d4b08588ad7e02bc30a244939b7b14f51a28389e8c8b2e236467ae1a327b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.26.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="b1576338af0365c1040d50e602d56b394b1fd5467b72a35025a410cab20ca469" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.26.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.3-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="980ff91a76061aa715a95a58c123fd23f86f118f0e806b0d3ef19f70f0143d4e" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.26.4" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.3-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.4-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.4.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.4 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.4
-
- Change Log for Release asterisk-18.26.4
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.4.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.3...18.26.4)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.4.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-557q-795j-wfx2](https://github.com/asterisk/asterisk/security/advisories/GHSA-557q-795j-wfx2):
-    Resource exhaustion (DoS) vulnerability: remotely exploitable leak of RTP
-    UDP ports and internal resources
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-557q-795j-wfx2: Resource exhaustion (DoS) vulnerability: remotely
-                       exploitable leak of RTP UDP ports and internal resources
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    pjproject: Update bundled to 2.15.1.
-
- Commit List:
- ------------
-
- pjproject: Update bundled to 2.15.1.
-
- Commit Details:
- ---------------
-
- pjproject: Update bundled to 2.15.1.
-            Author: George Joseph
-            Date:   2025-08-25
-
-            This resolves a security issue where RTP ports weren't being
-            released causing possible resource exhaustion issues.
-
-            Resolves: #GHSA-557q-795j-wfx2</changelog>
</package>
<package pkgid="baa14655c2cbbcb23c82cced2cccee88477ac3953b2a9e13e116230f841b630b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="920427dd4c837c19be0ec1584c5e3d1e46efc2094ebd5083a6d82f15a498eb0a" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c202f06e21f187e81a01957f9123f6c14738f8bc7ff203de5992a9fd3b28f819" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="736c6906f99155b2cb3765901933b6c976ebbcfcef22778eacfc1049823c1745" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="83d6328cfaeaca7fd51a94e24d7181427d047cc5f40be7c04f36900d9ea8e368" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c59a069881e49f3453218c6e75246a0ab1d1bf0fcae5219ae532a2ac0991b222" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="842eedb9d87ec386572b1b5691d91c533c17f3cc589acbb082b2c090d0d1624c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="fbf889db2eff8c82b7af52bffdb228a96ad2d2ab23750dbddbc5411a5ae33eca" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.3.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="66d127523fab3805b585dd4deb4758ad82c2896056494e55a66bd520f495294c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.3.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ebd5e4d57beb3da262d777692637e6faf7ab2dfa013c76795dad2b3dc25a3afd" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5aa2804ee8ef4e18c4a805ec3468aed17d4dfe1a3fc3147a4e746b8d5d97ab0f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="aaa3ae59eb07ed9a1a83987f0666e835115d16aa0d30881c10f5ad2ca59efd68" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="88919ba0b29d3625fcfaf5830f30aeccb1090846715319ca5336f016cd1b57ad" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1ab2d159beed79cc814f141541e0c28debf285366f374f0a8c06a47de221b1ce" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="a17e49732ed86e6f67ff14cb63ed1a8387914fc07d95542a9aa8836a54053d3e" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="41ceb7cc2c189337c7cddad571e9cb768418c6b689c241346cdc751859ba1dd9" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4c732eeef3a4f2aa1701310ae66e3b8884edc49197ad2247b91dd2711a29c01b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.7.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="34582950fc315c7eb6ac14c8f8a577709c833ad5300124d83f7b4b088e2cbc40" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896001">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6495843519750a519389da1436be848d3ca82d082a7ccc5e94be7dbf941a017f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896001">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 19.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/19.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 19.8.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/19.8.0...19.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-19.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="05ac17129e1b149de19f80fc3efded20d04c41b13cc489ab2a7309e0c50e6e30" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fe2c50e90517b9a84642488f2297fad673a23dea5f11f3987b231f9c6eb3ec1c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="490e84238da1c28d99fe27bfd609574ce2d8c6a0544a6956b77247fba8c76f7c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="490ec4e72e1430410bc1d2cb131f18435f49f0e197b8902a41c35cc9702efbde" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="6d022712814f35110f1188729331b652976e0503235a793efe5db027c0038275" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="49b216df289c25388364e91226fcc451cd509b6fbf56a98f58775a9e9f71a99a" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="dd22a011a528beac0c3ca298280ec04c0609e76364cd4055767ed39b128ac311" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="c44491e5249a673fb2859f00c92618db97ca31ebfc7e30e828a22b60e9642edd" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="01e3d63f90fbbdbeec2c7024fac40bda42ec7d0779ec185e5bc8e760cfa37ea5" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="d2271ac8de8a9d130f660cf4886b9ff1ece89976849d3811e611fabb8f28eeab" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="eaa2c639a8f5088acb9a1215668b0f82f4ced690ab4a0414a75ecde3b64efc3c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="533d962d3c91f94063bb7a911ea49c1e4e869a7fc8128f54c147f0ad60283b5e" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="53b29d979e69107ffb24d0caf0b6ad0900eec15dc36bc2dd0ed64aff5b8b6eca" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="6aff447f37ab1cfe08e287746c3ce839d182b9eaa3a5d244cd901594bf7c204a" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="d50c35241047be02e261d22a7374411d70c2a14f2bd3e225c4bd83e68cf22f49" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="3dee39c65a775f1e56061e7e75c345f47dd3bb213dc6496d2773cdad4ecad1af" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="1f700189c282eb877e67ff5bc3a340adbd9f015d3d66a37e27b046b018e8868c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="8ce67407ce8d4608a7547d4cc497523fb8a3dfafdfa880d9ad252ebca8cb6fb5" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="8e56315e726ad2ad0546bc03b2810c15fba765a60b7c3b0d2e140bc2536193b9" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="3ce96913d6449b77287df6df48844407c5123766937db8fc0bf079b3ab0e174a" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3a01f925cb39bd77420e056a8ccd69ce0d74a220eadc53006f057a4c4629ddd6" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1d4be4c89ca7f619c4fd1271cd0cf762052db70b013dced03dbd7b7bc7542f32" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="fe7afe61d5dcfd2e2a62d1a2d2e84fe6e62a27f6f4192ccad8b8ff99c0f58770" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="0b974b2458ec583a09bb0e415f9d96881cac053acae2d02a849a78beee498bb9" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
</package>
<package pkgid="cf9986f3e02e05d1cf906dda06aea77330ab487d021c4621298a1f60fd1b2054" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="0a3c8702629054098595a52893e2983ffc1eda1616cd487e08cb96aa6d6ad692" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="a8eb23390c2525d85d5bf53825f132ac2506214c2e6c862ecc23954faf209bf4" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="cd070c61aef6256a221a7c335d6be400e494e798654c8ef64f13306d69017cc2" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="3bb138098cad90a4e488246a7dffc6e41d4a92309f4fadb7cc87e0c62057d666" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="581c4e569681528d9051bca22404724862e9fcd1c6f39e46f33cd26b07b77270" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="e119262a76baf6817d0ceead21a2960a8e3a2a8804358f689819561bf9b456d7" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="a35e58d3e30f3dcb9cbead67d6e2e67adce73a960aaada571010ea639807bf43" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="089b30dfee42f3816fb1c16dd8bc4c731742340ca3a15ccfba180072610d5294" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="7dbc464a86af915abd30cd2fc7f397d29eabd5c1118ad36a7038b9351cd3e3fb" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="00b31df9fdbfed7e32ce20ed4d1ea016d2a8d1b340fe497b8f3ce4ea27095a6d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="92c7941059a9bdec1899dc1b98bca3c6d33265453f60f82daae0f00039c6d73b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="2e2e30f35ce96b365e0416bcf2017e6f4c0d355c4c314551a986fc4eb5c4dd86" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="ceed73db5583ebe8f335cabce2d8443dd4a9a761bfb683454121e1ceb528fcaa" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="39293d62d22f1317b8c42ece773a3120e190c1419d8e19a7b03695945fcc9af8" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="666547342189ae0c09f8a9f18230c665858ace90a4a7bc027e25862cc56212e7" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="4228280e2df079176ec4368bd0cfe2764b652136aaa9e81268114907136ea893" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="2feab30c46d38c473a2336e975dde1756b908a4d4ee19273c7731a5f2ad19606" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1747915201">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="3eca3d505b454fe8c597c40821c41b6d80549d0662033a8da4273670193aca58" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="391d67b5abcf0d10c5ecfff5cab67ea97205f11cd5c1fad52c345be0d0f8f969" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="7caf4174efc4bde7134477402b71f3b8dfa7ef7d64018b8aa594bd95e283ed72" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="c6047867a236dfb735520eece2c1a09042d4a6ea88ca4a7d057614bc6a5a12fa" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="b3a99378c43caf17229e9593bfab2efda7205a2a4fa42b7fada3fe091f25c02e" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="407935bd22b6e3a271e50d87c1aba2173ed4ec4cd06627d28b11184f26c328cb" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="4f3376c25b31948e165e00ff27ab97bb1db92223c51b435ba7975df875781b55" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="b5857970a720e372fb75141582bfe9acd90fb1c532bcc62e71afbec5cd8ae4fc" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="3dceed0c77a61818e586aad6faccedd2efd37b28ef8752e71e192f644e0f7470" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="ae351bc2a9b32a8689a137ef4916edb14c8d2be1f32699a5eb5894cffecfd89d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="1fe60091c01ad9e06d3315bdb203e07150c2ba8e449e31ac4117d3f7f6b8629f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="3a123d75468f7cca76c70bb82f914daa2213e0daa9fdf0f1ae953a099f12ea6c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="dfa876344bcf363fb458eec53002aa158dec564c63d3576f8342e5b1dbe9de74" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="b5531a3084ea8d7311070fd61c3bba4fbecc6527784c727eabcfd640e1bfd2eb" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="362fd7b3f692c89e86ba5ea49d476a37fc2d59c50ea6c2c4913f80bc38fa2c2c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="f91e5c6b4b15f7e63ccd157e81b1552acbf3dfea115801dfa2e14c300e423526" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="fce6fa810533a05dee892461e04a3a330668aaa60a304868d0f917775fdb5d40" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="94b480c73cb6ade65aba791ee7208ed742350fbb19f054fbfa762cedd8be00c7" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="8063c3610e5bcf1a49fb0563d37a87ede0f7e2ec25247e550036354c7d935901" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="d07f64d012619d3dbb240fa9505467b8991b374c271b928a9dd3200be244ba7b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="dfe05c7c697dd698b29cb43ac76f506ca2b77895a239f34028d1eeda8bc5a019" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="177d9b9a8ba6a5476fd8ec427079302c4dc56828f75ecfdb1d5812ea5e83a162" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="6ad677ad4a64e5752b824f01ebd375df4140ab6c85573d9b5ee375363892c9e8" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="aa3f87f064bf6e7c2cf0ec70b9660320362324bb475eba2663a2a284d01c51ac" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="4068c15860865301599290f2a1800efac2159d08fe962e8214e56dac50e39854" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="4e8afdb357f1a323546b78cd2ae21961bf7be312682503c773bc73183b80d7c7" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="deb5eeb83622e877c0054ef735bf4a9fd0b7a751d21f0b1ffff5ec848e9812bb" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="d0a12f52321c53185052e9f26fc4208e673b701b5523a818237de92db5e05cb6" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="cfd48ad50279a494c0cd5e5cc65165ddb8cd00b742b73ca56648db4cfedf7fe7" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="aeb47dee0e96b46f7045a761e753a5ab7fec0f09a75e714ecc1a4300332d38a4" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="99b1d6d4628b686770e9f3bdec13f24aeea855890c21572012ebb5aad6b605ab" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="0c04b8ad4acd2b41ce9d21ffb5b7b48bd7d4bce070d16e48148c02ad2d344ec0" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="5bff66131b0b2f5e78d141ecdfb31949941000f70d280df3735fa86e781f11e5" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.5.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="51a9ac63fd36f54755b54250b27c8504da92888269e43e8914fd7e6bf5c8f7e8" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="dc538e12a796c1071a291371aa059f2958fe520bfe422fedcf5b091601caf588" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="2ebe46d907388468f029c941705243b86c8331157332d3a221b5ed958a7ad1bb" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="70e1063588f6d6976489fe22d6e66cee7e4cd27e19f4c66c33dbdbeb89fa437c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="bcbee4163405bfdaa2df3efd2c7542c0351632c3ff676cd6376f4ff1689974fe" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.8.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="87934d934298c4a0424cef14c0a6e6939d36c23d392f9c61599ff4eaf496aa29" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="bbecb1cedfabcbb0ca7bfc8fce3416363771b50bc75c919fae501506316d71fe" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="b59adefeadcdcf1b64f60665de06fe7b82a1fd4261e0f6e54317971503f0af59" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="4c13a0c6510f7cf8d2f690d674f3400550f0b2bf06b9a54efb2e833612c62da5" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="a25233ff0834321aa3915cd4d09e49b8f48d67d0d5b5965770facff1a3078de1" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="af39485073d7cb0472df622f6abffe395615554cbc2e5d29bce2a71c4103b8cc" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.2.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="0fd99764abf1190fcaead037aabe8a204b2b89f3fc419287ca07c85f74bd8304" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="6e81ed685c34d3b0e3b9724adced133415e695eb35cc52c5e61183fca08ff4a3" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="11e922509cb5afc89fabaa0e34b9fd011048cba72e8775be4de4149227b89888" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="23.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="9c02273933bb3fe053309d4cbfb260bf5f6ce07d26344b4b5b990e47d23afbe7" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">* Thu May 12 2022 Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1
- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8615ed36bf40b78c6771e12937ab980cfad5f71abccad702fef1095652d4709e" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.12.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7cc095f37ae4f16e3c17afd01487248f200a602200c4ddfe9ce9f3b2efd657b7" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.13.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="dd9211527af159d0df18be81e5c778e0baedbfb9a719ca51aa16d46fe9f0ab0a" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="601f6f890e574c3f1ef664fec4d35601361e0c232c612d6e1184c0f2a8f8a62d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="d4d1c9b1aede09a29e999b8588718a6f1f67c236a709b9e714ba30aef6fbe06d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="f9f8acd8b6a8bc7cf11175adb7214ffc6d93e37100bc3332c319cfb4ffa41e30" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d94d50ca1ada37febd2fd93c14288a654b2a65626cb4e73a62d8e4e6254dace4" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="160a068241372ea91dcd90c75815765353b85f43f2093350be7b772c281e5360" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5f59676a0571ad89440421997fd11426a6b20ac3943e086dabe14d261cae3add" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f891f1f7d6e2ab84aa411131851004e8b3606c3cf1a640e429ee93cbf68d34de" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.17.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c43662db446230538d934b213c8ad59de71ed6cf2a5c93b1c58e107140ff7776" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
</package>
<package pkgid="ca47ed6962320d246ff576eebfefe15dc8f848fd4aa162e056049d50c580ce95" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.18.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="9ab2044a806dea5d9fece84e218a4337dff0d71ba63f8579aa8a21bcf0dfd7ea" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="64f8832a5f6bebc7dc7260437f43185a982e5afcb7fd4024f8281f8062acef9f" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="7be5b121120f15b870fe57d355c568fbc3fdf252e0ef2694dd28d938b095d9e9" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.20.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="c89c815339cb60fa473b1d3d550a286d754a8dba6a0ae0e41da2347234e78efb" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.20.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="92cf288874149ef5aa785895ed4a73f34a441937b29a00c306a37bf3405774f5" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.21.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="beb975350f12fe1de1a8d742c3128f7a54156b6cdccd2005dd95fe6b1835348f" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.22.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="4653cda066060cb56248136196d1c19df57a4cb23dbfa83608041ffc366a28be" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.23.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="083ca73ad194b50c476531dc45e7c9aaf383993df696604b9fb5038b696d2118" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.23.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="3a2b65d0f545a34365b1d80db67211e6a08de5abb20d439619e5ff904d7af3b3" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.24.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="9573f212535a6a6dcaf30d2d9592ce286576127af3ff8051a0e9327291621fca" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.24.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="66b9c457f65186c8e8b429028f5b7ed40e28fd6e690b657ce0f3b098da390cc1" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.24.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="a1463a46ceab4c062e80aac788c5037c071fb0d7b70450ef7a72451ab1aa86e9" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.24.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="2b7720a23f7d57a752571e7e8da97f2dd626d4b5db23ac47d7efeee9cdeddab0" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.25.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="b2fb9a0889a6caa79323a49998ca1c5011c7162f36aa52445ededf9a4cb262d6" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.26.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
</package>
<package pkgid="cf725381763fad32cb488d3dcc2a7b2d5d4390ab1c58c0a66e91a3a593ea8e4d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.26.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="7b9882727dbe1451427953b6d0fdfa3a70612aa67a851a8d59764332de22d87d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.26.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="6000bc803b74f5d22bcc73eae08b05418e58222768f5c112254c80f598c19d42" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.26.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.3-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="54347c3edbc1fbecf24ab4c7d130872d8fcdb70e2a42c4a69442199e977c6e1b" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="18.26.4" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.3-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.4-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.4.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.4 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.4
-
- Change Log for Release asterisk-18.26.4
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.4.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.3...18.26.4)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.4.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-557q-795j-wfx2](https://github.com/asterisk/asterisk/security/advisories/GHSA-557q-795j-wfx2):
-    Resource exhaustion (DoS) vulnerability: remotely exploitable leak of RTP
-    UDP ports and internal resources
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-557q-795j-wfx2: Resource exhaustion (DoS) vulnerability: remotely
-                       exploitable leak of RTP UDP ports and internal resources
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    pjproject: Update bundled to 2.15.1.
-
- Commit List:
- ------------
-
- pjproject: Update bundled to 2.15.1.
-
- Commit Details:
- ---------------
-
- pjproject: Update bundled to 2.15.1.
-            Author: George Joseph
-            Date:   2025-08-25
-
-            This resolves a security issue where RTP ports weren't being
-            released causing possible resource exhaustion issues.
-
-            Resolves: #GHSA-557q-795j-wfx2</changelog>
</package>
<package pkgid="52f3065c073d1ca1699aadf6ec258b88357bc536ecdc1c72ac9ea586e139243a" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c12ad252ede3d8801c02f9e8f80985780d4c986ba06eaa0b807b2f6d189067e4" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6468c13083b0f557e96bbc586c3fff205826f52606ec02e86a9c6a97189a3047" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="128cf5a9daccd023f29c585f6a496b28ae316127b3b329fd61568a05537e562e" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6a4524ab65867a968a922fda679f8bba1734edc6cd00a545a7bf69f0554372f2" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fe45faf98a3100b95d5d2d67fbf8a331f09f73393c2b2fe55623e0e6c7a58814" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ab79fe7d5e4a422fad0506cb21c15d5b3a1a4dbefd3ac3a88b2914ceb246d60c" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="7ce36a3ae00fe294176906aa31f88b47118e708f6dfcc45ba2e3631362370d2f" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.3.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d94657b8f4b183a81da713e17ebe14200886e5324afdfa39596cf7e7a91fb9ff" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.3.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="97481da270e675391a0136d7c2ee4755fae44324693143d0af12e27c1f19d86a" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e02e0a951854a328a25157438a3a3d1280805f605dc59985c6f30a2349df9e31" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="243ffd233beadee91b4e1684bf5c0386cc7c411c3898c35bac4687cb59441b3b" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fcf7826f67501e8ee20022addc0b09af479ebdb6a8fb594e980098dd2d7c37fd" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c07af534330ce41e5ad87f097ce1a7e451fa0f12cbd91a78651fcc3aabce5433" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="3aa6e6d017757ca1267d09e5d9ccd9705c0179a1402abca5662ffe6e81acc2df" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="f2d2c4b1ab241f931a3441e7810bbbea8379e7405ad446dbef1180f2dcd8214f" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b0e6a7f4ffa240454c044134534fe87351e6d7e60ee90680db631659af37a0ca" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.7.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fa6981c8b3a2ef3171016a1e785507f5743ca038ba7efa1e344c17670b013ae2" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896001">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f882f9e414b48b854f1895463a443603323b72c2c1765bc94e9f9158cfc2b18a" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="19.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896001">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 19.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/19.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 19.8.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/19.8.0...19.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-19.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="62f45afaad24b9fcc25b081d2aaa0e04f63233e97add17ae39f3bf8cb481074f" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9c30bea1e47ce04fd7ded4046021584fad6f67c64159bb10efe12cca9e16b3fd" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9c471563c4c7e5531175e102863e85e458a4c8816f55ae980231c3f0efc18ee3" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="495cfbebf1852c0531112a04f5b74dc80a5949e8c1bea0c32a8b13028b478405" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="6c457dcd0f5c868a551a27c57485cda25c783b2c7bbc0db032b4ba8f953513f8" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="d5ae488df4fc954487956926acd59a6753289bd76e017abc99212020866af2cc" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="38c4bd3009149fbcff8d1657f221bdabaaf358cded71fe56ec41ef5e4dea5f9e" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="6627701a99c77d8b7aabc3655dd4c0ad70c1956f36eac5a79be110e68df9673c" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="26e37e58d2b89b0536cce53ef8dcd39f66365966151ffc8ab2eb7db28fab6f5a" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="5484119db99fbf3a63a1763cd6660fc2681743a548e7808d45f05c415b9febce" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="be6111cf426df14a1de38bd0a0b2a52aa288da053e1cb82e7c15f64eee04fef5" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="4daa890925b45026f2bec646a3598189dc78b125aac2b38646c398245c26dad9" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="a2eb52311518f6f07884cadb524f27a40fd5524d118ae8999cf4ee14fe36b9b8" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="79f52a22e1f850c0b271436d82b2d00a620434599cfb75032798a5f013cc3ec8" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="c22ddd147651cbd30c04d964c5b69b212d2f3476006922521375532b5b5e6929" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="09831b34337beb6bb70ee8d8493164447de6e32a678909161bbce20021422190" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="868f641c65985a600da8595c4fd2227064d42e87aa57f2bf98de5aa1932c59ff" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="04bd4b1da1b8f210703854d1fccbdf915f8d6873305262e1ca83f7db78da632d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="92ec3099c872a79bd92a261ecc1b1a967a2016bdd07c81cc7df5f5aaf6d94b70" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="4b1bd69d75347bed36b0127564f22e4c3568d5bdc71eaa3bd20d292a6aa7e821" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a6af18a21b3abb1a9c0a19660a395e8065543589903b11c56651c28c2067910c" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e869c096812f55072cfc02cb32ad93b9c71bc8c2f7a2f23275e9b378a89a4f82" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="50786f08634d34e33dd7ef9afee3a76bf9d702dcbb04459f62e4bbcb8299aa7c" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="d1c8202210a3479cb088befdc3e9eb145914ecdb5bb049e6a32294c06911b80a" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
</package>
<package pkgid="65e29895f199b8d7f43f616ef51ef44b4a344e2e6a6a528117742a76cf45e93b" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="0ee0190d6e4450a6603b799caa02e275079a703ac1d8881c9fa9956ba49d4c63" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="f88d8c50c8c8a6ef84f508dd3fafe5bd846a046069cf01c4ff20acf48b4b792b" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="5cdc24a5ea0fd181ed4a8591999e3a7bd417404d602030a5caaf41a58358b02f" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="55d723efadb78211d161fb14638e6362fd44690861e408bf12fb839057ea409a" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="ca063d32fda889a1f020d8c7b8beaa6a18c4322c705e959fcf03ec50dce1a729" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="0254088ea9df3d101c65d25f80dbd63be3083629c376df79f0f8779c8fd16781" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="065279cabb2ed67fbc200cb46228ebfeefcd576c124da45f7f35b78c1964f4b9" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="e76154566283fbd292fd351817e7bc771f838a7e3598936294af168e481cba84" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="d6381ec189a41c14c350e0b54ce35375f447e38c73033a45900fb27a747ceca1" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="21346d1a7a34fcccc823a5227c7203a8927c7b9155af205b367e761bde289e31" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="54ebd512ed47a1500885c1ae45761f33b7a96a34d5d5d73beca629889ca3e05b" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="02250bcf8ee9894b8707f7623e1c3b155197c339bf80e73d656c58cd2aee80c2" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="6f301d9fbb3d38e3d874d2b6c04983b93259a2dc1aa49b9f204fa99878ada8cd" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="03ddb544c773246d7b2633977feddfbcf0929b3f1a07b3b79c4cee388f3343b9" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="c98c19d0c392f9087f89981c93fe395a75625307664ac42344cd8c0d35a07c19" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="17d231d90c725b5979c9e50902207a633aa6cc3b9ddb8d9c307d346d5e9fe8b4" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="e28541a06a33a38133e099b8d43faf51957bcb209557efb43e07e82828125d96" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1747915201">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="5d138f1030e84d693808ef9b534d8dfdbfe250ec7d911e29e945936f4b184eda" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="a5a3bd3664025c748ce931c2060c651185b706cd4c7b1b339e68c8f0ab65c01c" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="e226a80d2cba224765e15b35ec15aec47593cfa41922bd949b88b3cb599fc5b1" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="68f019ebfb702929928898d665eb2326c8c72f2152912605c73856fd60f9d4b7" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="be6334cbe85751106ffb0375c14e2a653d39d9035f4d330f3117709c74d9a8ff" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="111d9e62497a075901c55f41243b83d6a8499411003d5095cf37f6a02b9614e2" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="8a9856c6cfa409f87f571f98e0f394cbfa63c09e02b4d4c0c29f941c502e1f70" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="d81a76aa57e99603afb021d2669564f15a64520c8849ff8a89db89fccb907884" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="e3bb7074a5624cc36db32bfcff0594c5e3bd51cb8a0374c376ec08ecfea2d839" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="e7533313d4e590f652a746febc4f0294697950c8049149da9f954605e6d8e45b" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="1d51c6c02f5fa0da082f809b668f5d71dad66da15b35c6236d5fb6b7a12d94d7" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="1e59d2dfc87c983dd8d0c4b76f3ee2eed28c21b395ce93b8a2d81acf579ff056" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="d2353980ed9db6be33d49afb05a4d84a370947b84ff172c9addb4fad6d64dad9" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="cb8cebb50813075757c46703d3cfdca5032ce5bc311e21acd1b8a624557dffa5" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="35ebc4824a2abfd9d2f134920161053017865af5688fbff42638341f3710dc12" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="9d6f63494be51dc63baf300cc2d00084935aa967b756c89ad907d524e1aa2d90" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="f6c2c135b7cbb09e8e9923baac53b30fd760fdea84938f3891c435361be53c32" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="dafc319d3019de01c5d3971fcb6e7c90bc6d7d6a820fd97dc4bc8c29ce324f51" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="d9e358135dcaf399b36cdec3f055aae4d0ecc2d0eb4133c57ac33144c8ec1dc1" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="cd4f192d0c2f03e2b74b8757ea9ccca429203b42758fde6b2b3676a287895f02" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="7a52b9ffaefabd5df15be13257d1e01f8bd56f65de9c82dea76f47eae0446524" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="21.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="50ac6dace89d5d844c9767c4c7643a304ab887c95dbc21e66afc5af189629b40" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="7b53aa12ffc79c2386385929cc66e69088aa5d0b9373ae04520bcb90af488e08" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="ffcf8178f083d234cc702630d9e3f74a6c73cd90038dedce989ca517a6b4fb20" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="a754ca6458a295ab12d891daad5d9fdfb45fe19af90c62561c20b0a722430ee4" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="a99c597c98fc830c28a7beb8072c5c13a8f76fae2e07d2606a7a8a9486191f9c" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.9.0...22.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.10.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk 22.10.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.10.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.10.1
-
- Change Log for Release asterisk-22.10.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.10.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.10.0...22.10.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.10.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="ac87c1e47347a75d6a70b1f229f10076f31ce522afd2d52a729b61f23020d6b8" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="60c1f9781a761b1d05648fc1871fde74b29bdf1b1f2ab84a154de0dfc4de9554" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="9e754048329b00fdad6f98a3acd3ed426596a0b353e505ea01f0be10b0df8d78" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="d2d8890b11e8d3cb2cfa7f96b9b205a03bdd4d77801f2028ea671b5b013757fa" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="9b9fe118e7c3ad0f6b9b1ff3ff13a06bbe81f76d22f7e2acb71f08f17b1fcc05" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="9faa10c336488d8f595771a22bdd24f573674c014784eb0031851d1120050c2d" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="b1e4007ed9a91d7cc9c8ec75ede78dce7d31a9b06298a1fa370083b87ef0b683" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.5.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="37d8c5888cc7faed8d8e4be4ebe206299352a5a81a41fd9dc84006890ed48bc1" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-            The new "suppress_moh_on_sendonly" endpoint option can be used
-            to prevent playing MOH back to a caller if the remote end sends
-            "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- app_mixmonitor: Add 'D' option for dual-channel audio.
-                 The MixMonitor application now has a new 'D' option which
-                 interleaves the recorded audio in the output frames. This
-                 allows for stereo recording output with one channel being
-                 the transmitted audio and the other being the received
-                 audio. The 't' and 't' options are compatible with this.
-
- manager.c: Restrict ModuleLoad to the configured modules directory.
-            The ModuleLoad AMI action now restricts modules to the
-            configured modules directory.
-
- manager: Enhance event filtering for performance
-          You can now perform more granular filtering on events in
-          manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-          = PJSIP/` This is much more efficient than `eventfilter = Event:
-          Newchannel.*Channel: PJSIP/` Full syntax guide is in
-          configs/samples/manager.conf.sample.
-
- db.c: Remove limit on family/key length
-       The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-       removed and will now accept families and keys with a total length of
-       up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-       dialplan applications, dialplan functions, manager actions and
-       `databse` CLI commands. Since the media_cache also uses the
-       `ast_db_*()` APIs, you can now store resources with URIs longer than
-       253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="d2c8b824d1a6e414bf05bf8b5b73a5b88b24dce696b1671af2922324d452e3b8" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="56e746a6fae68837e18b8a63c2273b66407e7c011587ccd82ba94fd51ff521ad" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="3b7379b623765aa9cafb30ee72a65f19df12a92ee64ed442c3d5bea4d0f6baaf" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-                
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="4de036b87a7acd1c87a1003e1ffb2f9eba120505a394299ff3822d8c0aa8d6e9" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.8.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="e3e7ffb84dbf61f0aa9a36d8ac4cfd3e0e6016c125b67f31ebdd09d47b708dad" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="22.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.7.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 52
- Commit Authors: 16
- Issues Resolved: 36
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.7.0...22.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 19
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.0...22.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-   1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-                when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.8.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 22.8.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.8.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.8.2
-
- Change Log for Release asterisk-22.8.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.8.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.1...22.8.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.8.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-                        Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                        potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-                        ast_coredumper running as root sources
-                        ast_debug_tools.conf from /etc/asterisk; potentially
-                        leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-                        The Asterisk embedded web server's /httpstatus page
-                        echos user supplied values(cookie and query string)
-                        without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-                        ast_coredumper runs as root, and writes gdb init file to
-                        world writeable folder; leading to potential privilege
-                        escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- ==============
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.9.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-22.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.9.0
- ======================================
-
- Links:
- ------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.8.2...22.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="1cc155f41dc6f3e00132ffa20cc1f03da45a8e9de9c0767e7e9a82de0dd45291" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- manager.c: Restrict ListCategories to the configuration directory
-            The ListCategories AMI action now restricts files to the
-            configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-            Author: Ben Ford
-            Date:   2024-12-17
-
-            When using the ListCategories AMI action, it was possible to
-            traverse upwards through the directories to files outside of
-            the configured configuration directory. This action is now
-            restricted to the configured directory and an error will now be
-            returned if the specified file is outside of this limitation.
-
-            Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
</package>
<package pkgid="273d269c640c44987e1525c0cee35edd86488e682e9c1dfcb318f358470bee2c" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- sig_analog: Add Last Number Redial feature.
-             Users can now redial the last number called if the
-             lastnumredial setting is set to yes.
-             Resolves: #437
-
- Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- res_pjsip: Add new AOR option "qualify_2xx_only"
-            The pjsip.conf AOR section now has a "qualify_2xx_only" option
-            that can be set so that only 2XX responses to OPTIONS requests
-            used to qualify a contact will mark the contact as available.
-
- app_queue: allow dynamically adding a queue member in paused state.
-            Use the p option of AddQueueMember() for paused member state.
-            Optionally, use the r(reason) option to specify a custom reason
-            for the pause.
-
- manager.c: Add Processed Call Count to CoreStatus output
-            The current processed call count is now returned as
-            CoreProcessedCalls from the CoreStatus AMI Action.
-
- func_curl.c: Add additional CURL options for SSL requests
-              The following new configuration options are now available in
-              the res_curl.conf file, and the CURL() function:
-              'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-              (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-              (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-              'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-              (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-              See the libcurl documentation for more details.
-
- res_stir_shaken: Allow sending Identity headers for unknown TNs
-                  You can now set the "unknown_tn_attest_level" option in
-                  the attestation and/or profile objects in
-                  stir_shaken.conf to enable sending Identity headers for
-                  callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- alembic: Database updates required.
-          Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-          authentication digest algorithms' 'res_pjsip: Add new AOR option
-          "qualify_2xx_only"' ...have modified alembic scripts for the
-          following database tables: ps_aors, ps_contacts, ps_auths,
-          ps_globals. If you don't use the scripts to update your database,
-          reads from those tables will succeeed but inserts into the
-          ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="505c4d84c6e2ea0c1073204d7124c7a04a29ada2b1a441b95bc03100ae13c1ac" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="b54859f85d20f0e61d6c7f583125f05d653e6d7cd78d454881cc02440488a6b9" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
</package>
<package pkgid="98fb844c48f5bbaa320f3312062e92a2fa4717f2f91562452cd39191268e8857" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.2.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.1
-
- Change Log for Release asterisk-22.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.0...22.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-   Using malformed From header can forge identity with ";" or NULL in name
-   portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-   cli_permissions.conf: deny option does not work for disallowing shell
-   commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="220830180d9eca7e454da1c9620fa2096254cc54ec719195fa2b243c28f4a5bc" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.4.1...22.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them. Full details:
-    http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="9e56a68fe84a029e0f48c4529fe953fc914f79ffd565b0d3cade3d0e4197043a" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="1fcb599fc4509c2e3d9d5fa7d559842ce36798079a84a6070a5106ebb68b3765" name="asterisk-aeap-debuginfo" arch="x86_64">
  <version epoch="0" ver="23.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.5.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.5.2
-
- Change Log for Release asterisk-22.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.1...22.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.6.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.0.0-1" date="1760529601">- The Asterisk Development Team would like to announce the release of
- asterisk-23.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 45
- Commit Authors: 14
- Issues Resolved: 36
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-                         A specifically malformed Authorization header in an
-                         incoming SIP request can cause Asterisk to crash
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- Upgrade Notes:
- --------------
-
- config.c Make ast_variable_update update last match.
-          Config variables, when set/updated, such as via AMI, will now have the
-          corresponding setting updated, even if their sections inherit from
-          template sections.
-
- config.c: Make ast_variable_retrieve return last match.
-           Config variables retrieved explicitly by name now return the most
-           recently overriding value as opposed to the base value (e.g. from a
-           template). This is equivalent to retrieving a config setting using
-           the -1 index to the AST_CONFIG function. The major implication of
-           this is that modules processing configs by explicitly retrieving
-           variables by name will now get the effective value of a variable as
-           overridden in a config rather than the first-set value (from a
-           template), which is consistent with how other modules load config
-           settings.
-
- users.conf: Remove deprecated users.conf integration.
-             users.conf has been removed and all channel drivers must be
-             configured using their specific configuration files. The
-             functionality previously in users.conf for res_phoneprov is now in
-             phoneprov_users.conf.
-
- res_agi: Remove deprecated DeadAGI application.
-          The DeadAGI application, which was deprecated in Asterisk 15, has now
-          been removed. The same functionality is available in the AGI app.
-
- res_musiconhold: Remove options that were deprecated in Asterisk 14.
-                  The deprecated random and application=r options have been
-                  removed; use sort=random instead.
-
- app_voicemail: Remove deprecated options.
-                The deprecated maxmessage and minmessage options have been
-                removed; use maxsecs and minsecs instead. The deprecated 'cz'
-                language has also been removed; use 'cs' instead.
-
- app_queue: Remove redundant/deprecated function.
-            The deprecated QUEUE_MEMBER_COUNT function has been removed; use
-            QUEUE_MEMBER(&lt;queue&gt;,logged) instead.
-
- cli.c: Remove deprecated and redundant CLI command.
-        The deprecated "no debug channel" command has now been removed; use
-        "core set debug channel" instead.
-
- logger.c: Remove deprecated/redundant configuration option.
-           The deprecated rotatetimestamp option has been removed. Use
-           rotatestrategy instead.
-
- func_dialplan: Remove deprecated/redundant function.
-                The deprecated VALID_EXTEN function has been removed. Use
-                DIALPLAN_EXISTS instead.
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- Alexey Khabulyak: (1)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (2)
- George Joseph: (7)
- Igor Goncharovsky: (2)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Mike Bradeen: (1)
- Naveen Albert: (23)
- Sean Bright: (2)
- Stuart Henderson: (1)
- Sven Kube: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.1.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 17
- Issues Resolved: 37
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- app_queue.c: Fix error in Queue parameter documentation.
-              As part of Asterisk 21, macros were removed from Asterisk. This
-              resulted in argument order changing for the Queue dialplan
-              application since the macro argument was removed. Upgrade notice
-              was missed when this was done, so this upgrade note has been added
-              to provide a record of such and a notice to users who may have not
-              upgraded yet.
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (2)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (18)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 58
- Commit Authors: 20
- Issues Resolved: 41
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                   MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been
-                 added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (4)
- Paul Donald: (1)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.2.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.2.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-             Author: Sean Bright
-             Date:   2026-01-23
-
-             Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.2.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.2.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.2.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.2.2
-
- Change Log for Release asterisk-23.2.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.2.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-   [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-   Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection
-   [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-   ast_coredumper running as root sources ast_debug_tools.conf from
-   /etc/asterisk; potentially leading to privilege escalation
-   [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-   The Asterisk embedded web server's /httpstatus page echos user supplied
-   values(cookie and query string) without sanitization
-   [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-   ast_coredumper runs as root, and writes gdb init file to world writeable
-   folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.3.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-23.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.0-1" date="1775736001">- The Asterisk Development Team would like to announce the release of
- asterisk-23.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-23.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 23.4.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 23.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/23.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 23.4.1
-
- Change Log for Release asterisk-23.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.4.0...23.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (3)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (3):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- #### ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="8e61791fcb0e9b55739133fe9ec420b32e75bdc75f978fb75a19c55d5c6e42a2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.10.0" rel="1.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="562446d2abc3ca73d4f04375a1bd2b72e980a4b2203098ab32f2a5f35afe4f5b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.11.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="76663a9b5e12a330549bb4c635a2a4da1fa670d79dcd1e874f6cfd576f19376e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.11.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="754b5be94938afdc1db4ba111dc0e7753d8fd09f0acd4882e04ce9d32a4321ee" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.12.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5da56cefff33fbd956afdd004b87c9b07d1865a5f55aace4a1cab8ebc7e9f305" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.13.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bf3ba14f87735fe59593f42dac672e502db9c81d05bd25cec30ca6689ccb6388" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d4c1b87670439ad5b3e842bd75c723d0967a87191e18dce09bcfc0107c06f80f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.14.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="59f80edde08a8c376774deabb297f5a7b18ea9f8e011038f5bf8e5080dd4e67e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3a5642c800eb7fae9a0a91032a82d8767bc1ed0cc18f885a3067f1284ca656cf" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7d6a9cdc7f83e5035b6625d849385ada71bdf4f869d69e763c5360615c60dce4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2a8975292820213e64f31f4c0f5db0ab7197685daab973f0eea9950dba159e57" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.16.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9568c586241dd7f284f3c657a0f98e57e728582c125262a4db11acdffd5a633d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.16.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fb001e4675470252aee8515114a059741811e066e90d62278a894af9c1dc8f52" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="94b0dd128bc7e86d79dad8768cb1ef0db6dfbcb5cf5dce1f258ce162b1bd90a2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a7a8243a314deb6263928fd8f9456716b974fc479aec54c93889f243bf642a44" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c4e286655f2eadf64e159ae47ef3d1000abfe5a2be37acc7221d24c1938bc534" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.19.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a86633ebabbdf2a2e455e152762c3563a85edd694f2ba75461bb66b47ff539b6" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="75174248bcc5d6599278f827faeaa3f398b58714e5fce571d73d5ffcf58fe0cf" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.20.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="65541143de71dc174a27e8999ffe707673c38d2a92669e18dbb98b876a453cb7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.21.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e005c6c1b662c5f4734129b2f1803939fd35ab659812418662622e31225ee831" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.21.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="945eafb6a6875fbf9de2358027e4bd73ccd555e450c7c089adb39633d02038d1" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.22.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f3b20047ce353bf76282fa571ae44281ced8a5e5c8d0fd9bc112b8a0204308ba" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.23.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
</package>
<package pkgid="edabe1fea754f7b70dec79a24537c9be963b1539f199c21b954690fd93ccfe79" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.24.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ee6ee833f636f6eb2105f88849411abbe3fdf468abb66bbff7a5daa2acaf847b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.24.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="88fd7c11c8b916e20c853a75864c88b7d191b98d4d66c030ac23ee12156c6695" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.25.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="59083b5fb4662ae6f232cdfa88cfd53d04c38b5bdc21e8fad7fc99dbe41226a0" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.25.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="50a7a49ea264f5c2d9d51ce7e20b444f85426013dc23235f0b1a1303ecf335fe" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.25.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1d0971ad136deffa1b6351354f31635b81fa226805914724a8a941752504adf5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.25.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cfe49165555e84d870ade6515084a6d1073df3b4dac7011176e51251b6c7c00a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.26.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="639bd1b13acd3b85f4c5e873c5fa4f1e6e77f4a6ac752c6a410e0a7298a09936" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.26.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9b34d0781a3f90e5b61199be4337add97b8c2cad2817f1a3531ec29147e0ce7c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.27.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b7e0355bc660d3381e395841d4a023812a7550afc1707569022a284cde1f270d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0023004394ac360eeaa49eeb1f17ab93c4e64b07b559d265304675d194b0feb5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="a93b0197b1b81a1035921d10b0320132d8fd063870f3131f05bd92010b4c88d7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="394b4f0e973c6dfa1e7254c171992bcacab94c94775f9d4961a0354c711535fb" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.29.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="71b73856ccc8f79d04a81f44cbac86b7ca06b833ab632260bb84fa2e45e532d9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.29.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f102fb5078a95ad50678c1daaf88f1f3ea846e8b288e521173d4cc3e346b0dff" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.30.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option &lt;E2&gt;&lt;80&gt;&lt;9C&gt;peer_supported&lt;E2&gt;&lt;80&gt;&lt;9D&gt;
-                    (Reported by Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn&lt;E2&gt;&lt;80&gt;&lt;99&gt;t be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: &lt;E2&gt;&lt;80&gt;&lt;A6&gt;may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2eda46a2ba18ed7d7d789143fcbdf99b5215084ec0384006a096e7047bede333" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.30.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option &lt;E2&gt;&lt;80&gt;&lt;9C&gt;peer_supported&lt;E2&gt;&lt;80&gt;&lt;9D&gt;
-                    (Reported by Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn&lt;E2&gt;&lt;80&gt;&lt;99&gt;t be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: &lt;E2&gt;&lt;80&gt;&lt;A6&gt;may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 16.30.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/16.30.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 16.30.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.30.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/16.30.0...16.30.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16.30.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list
-              before applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="68dde42aaf66848824480f05c7f1f23f77f4ab5ad154b1a1928555b55fc2daab" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.6.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.0-1" date="1550232000">- Update to upstream 16.2.0 release for bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
</package>
<package pkgid="d072905c802d0fd3222986ec7532fd7ec16f613b7616e6a6b5da193ffe0a5b14" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.7.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="825d332e78393c849ace46cd620ab2b03a5f7d013f26035281bc2b6988d91911" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.8.0" rel="1.el8"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6654583cbadb6aad8a7fc8d5a213535f5ea62c719b39fae0db4238d394cdb83b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.9.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl) 
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes) 
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9cc22752ea01649c637b7f0cc3d224012ed34b8129fbaf159d0097999ebfdd40" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.0.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e9c23e23df0b27ec8d1e0b2aaca90aa415dce5290d3d2bf5ca9b01d5316d3954" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.0.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fe5fb959089755dce5d1ef6755e3c4cbc2632b7472ef648a872adfb4c1795597" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.1.0" rel="1.el8"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f6d8efb96e8112ab33d3a620b7b0f049e3831c33c1c34d00fe881718b71691a1" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.2.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0f3faea8edca5459156a13237758da20372e12023db533dbe59af2bf863176e5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.3.0" rel="1.el8"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="78c3a87e561dcc6ec91616af62aa913305871c890aa583f224c269e8e0e64776" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.4.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c3e53d786752c8f6f86a58666a3a1e5ee72facfcfbe4b2bb80c1430aef401043" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.5.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0ba51e4b6250eebe19f944edc6e4d58a3311792f500cbb115065b9f5aeaeefcd" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.5.1" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ba16b5e982e34613c3fd37163371ad6ab4a61eddde9681c1cbff704a99bd6373" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.6.0" rel="1.el8"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="32f48197c753cd228468e52f45520084755e8e67c1347b122547cdb9ace5b763" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9e97f1d82c998500c8f62dcb72748a9f1970bc2364e03bbff8f6817c0f71a130" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9498d850bcf8efa1594dc6f85cf1ddff81542dddf68fb71b7253fddc77f575de" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="349c77a52a8e50d7f891a74fa571e7bc342dcf407a3c4c1e23a9c6111e08a4a2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="95b91d7a1d54b79175e8cc92484f953df614dda13ce7abd275a004783c381afa" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="75666e1778a97f57b687d609c4a6afb3b3ab78fcd890a0a90a5d168291deb9ec" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="08af5111a732044f130d06ba901c6b137a165e7ecb387f39953077da89f76c68" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f96bfb69a517b76c9e2c73d1d20db793fe162da43c6ddf4d565f9f6d81450db7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.4" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="66b46d0443c42e14b14f5f5567a2b33d2ba73fe6d4d69401bd5e8ec53b7a824f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="20872abeaca5c6811618b24e847d95fcd706ab218ec0b31f4e0a7ec2e9372ed7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d8f1043b1ac149a042c947dd09b9562940c90ac5055da920b63834640fa4bf38" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2b8a827669a25ccbea9acca8c04d76dfe5bdf214dd6d367f327c3658134484a5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.1.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="67fa72c935c38f847b7eb608a9baee5f0bcb4ee87ffd092689d94e3ab8537c45" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fee6b4e3c3092452f2fdf7736462a059d51134d7ec3a1cd770bb43f07f542914" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="adc4a621c7a368e29d645abaa9a9d4f2030786d53d5b321c51338ef3d739dbbf" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fe92fb5a5c6bc8d7bbca6a5f711ee19083a2b602eab88bd4299768c224245d7f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.11.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6913f1c520debc807b10e38fa1d8f4c83d0e656986d8776bf498dfb2f75c6f0d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.11.1" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="63be4b2987d7909bf2d4fd23d7ef4570b9dc7b5609d2e49d8a9d1b47dd9fba6e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.11.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a50e8c21524292cfff9c1670c595549461e35246a746ac687fa87ac4f7c8a2dc" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.11.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="38dea4f6d80a2eac029f3e4c08dbe9331d19f46ed8a8fb7c6befc66b6e813a56" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">* Thu May 12 2022 Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1
- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="eda73d55e8841cae7b72adf4d9986f4bc0c5f02ffa6ed360eae6ffcc0af220b9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.12.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1f3d488b84c36ee0135b444069c62174bbee67cee76817174e1d41cdf164e133" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.13.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f7f5b607cb9a0cbd8a21cb0c462ba6484240e1e349756e14f0432811ca34cbe7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9de03401416da8e5badaf5b252eb6870343f54b92e66cf5c5f3c4be883a477f2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="4265d2f5eb9b4d503826c9c2ce60717424aaf9d5196ffbb5310bc3f23068207a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="27b6b2e36622f9c68f7c026a1ed7e840d645f8c4db9b5081181ec99195560edd" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d9b1fa61d6af440ecb61f6b4a71cb9f42f71f79f09d6a40b2fb4b5d16355631e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1dd8b1e5367735b2733ed8cc36e8bc9d340d3608b7ced1abed945c3530aec3da" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fe06d836b3da7cc29192ac872b0bf6ea5efd941f30cd70d45749c75a4c37fd7e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b0b0361665ac83018123525b9de6e0d1bc162b9384440fe63e8a6dd1a2df5973" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.17.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f2df3b45165a35d7ec6a234f3558eedf5aa14291f24cfcca09d1c3ba5939a449" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
</package>
<package pkgid="86af8bf8e37e7bbcd01efcef9af75762c533b6dc8900e546236bb1902ca57886" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.18.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="d81ab9f5f79397c1e861bbd2543553f6999ece17e2db2398ef3bb4f2d7275707" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="d77117824efaa6b971ac46008cae769f8366b3c869c2ec44f77391d0449bfb7c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3da7e567fb31f647b1632c08149e5407315988cdc9a1ca7d9d71410d24232b39" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="682c8c30b825500b7b1fe92cd934b2089e0fabf1f6c63be141e0554f6ae49fa3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.2.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c4c9370ff6b21200c3131cf5ff4cb03344a21d23c720af514e42d02be87809b7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="15d5156e89417090d14cebecdf73ee99d18ffed3b8b5d7f033e4f2d32cf55a1a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.20.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="bcc37d99123a3e83292debe752eab322b766d9de9e944e7a867b6cef2cb0163e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.20.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="1a7403afd98e5722eae4c6d1ba90dd66750efed29f43c7faa27ad65e9d392115" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.21.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver - add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="8cf9516b65ea91ab4cfc6af69693be8e37d1d50b6284ea79ba61826627193ec4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.22.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="0d2e9cb323afe89b3ddccf04b8f6f7ec67be5a0505718892622d1f93d47055b9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.23.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="3516c5c8e4004c43426e94db00e36ee6deba7c9b8d8530d98af396351b6c2a37" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.23.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="a4bd2580e395df6e3adb0a1f74a9599d18d36062411abd4d81eacf8de48e8c96" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.24.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="9b6b2dd54b054da4051d52ba74c1618a822ae3a64a06e5baccb819c997f0f851" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.24.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="c2f1d06d0303292feb081cf995261c88917ba8494eeb29b1ebae927e653f68e9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.24.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="6cd19f960e45dee564dec2cb1e0b17aea78d86b349d03d0f6af8cef47cdd0f27" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.24.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="647f6fac1fb099ff178bc14ca8954e419496cc45c3381cc54116e98ae783d0bc" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.25.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="51242193b64623406cfeec1d09fee6126bd5b7ee27730b35ed92a531ee7f2bdb" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.26.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
</package>
<package pkgid="3cacf0955cad7c2ac63cbd4192c3abd59fd0e9d20a4e529812ad999475c91b16" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.26.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="70fdcbc8c2ce8c5fbb1c5888ae211c0461f27060d1c74769905f8d8ff81d9470" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.26.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="0b362fd2cf79d4be07eccab1235495aedab9497db89aa9533ebbd5bb4e959b28" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.26.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.3-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="a5ddf246b9a835d7719867f4a0a79b7ba1f3eb1fdab1003ce48dfa70b205f2a2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.26.4" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-

- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-    [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-    res_pjsip_messaging.c: Mask control characters in received From display name
-    asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.3-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.4-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.4.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.4 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.4
-
- Change Log for Release asterisk-18.26.4
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.4.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.3...18.26.4)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.4.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-557q-795j-wfx2](https://github.com/asterisk/asterisk/security/advisories/GHSA-557q-795j-wfx2):
-    Resource exhaustion (DoS) vulnerability: remotely exploitable leak of RTP
-    UDP ports and internal resources
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-557q-795j-wfx2: Resource exhaustion (DoS) vulnerability: remotely
-                       exploitable leak of RTP UDP ports and internal resources
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    pjproject: Update bundled to 2.15.1.
-
- Commit List:
- ------------
-
- pjproject: Update bundled to 2.15.1.
-
- Commit Details:
- ---------------
-
- pjproject: Update bundled to 2.15.1.
-            Author: George Joseph
-            Date:   2025-08-25
-
-            This resolves a security issue where RTP ports weren't being
-            released causing possible resource exhaustion issues.
-
-            Resolves: #GHSA-557q-795j-wfx2</changelog>
</package>
<package pkgid="bb7da253088737f285ed2d41dc1a5e4f423283145b22ef6a7805cf7ee05d6e9a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7bddb9a4eeccf8141740a98ee11fecb21bd93bf43a43846438d0d603ee1bb6a7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c8612ddd918723ada94bb61e684c35e6f082f57a75e3ddc84aa565d233893dd4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="91b830e1d8cba27cbd1982dfd04abe4e858abfdd5df9684c664760152b216119" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ac49f4bb58b9ff9936d30629f4c2dfa302e45a6f5c045095d80c6710a362ccd1" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="434a603b6f77a3347323b4dff425a47283669a5bfee0e32c00919f7b9562d2b6" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.6.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="f8d8ec31a61659b390719557d38f9c4f3410ce74b6c48b360fd862b05284845d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a64444cc8cdd725a1b76b9121d78f4d304f772a37b09fd9c3f60ca196ccd83c4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.7.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9ef0fb26a9653bcb4ac6b228be4e12839f3ef27f18e7e36acfc08721711bd0c3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4151256dca6333eea45d77b597f0cc44fe2a32f4779cce1f5ec383af377bc0ef" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6d24c7a2ee7ca94667641d64cdd6d581665ade790f2d62747026550fbfd87041" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="17f56889ed9b50a25deded62c7cb5eaa63573279e780dc9182bd607362edb582" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9de74dfff8dadae286fd2dcce77434e70efb2c741a69595901a0e99f3fd879f9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="93e3dfe1fdc9d5e6bddaf8224fa5a99b354f3f1da59673e3ec66b757d208a490" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e8aebefa72230ed82e06ad4cc03ae2fafb485fccbd06ffda3e60c4cda4dd4d3c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!
-
- Added OGG/Speex patch to avoid crash when writting speex file formats
- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9cf531b09d4b67deb38797fb26a5db649946c1a94f628a37247e9783256d1ab7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d67e1b74fe9fad6f8d7e096cdae68347babd4a264240d99c313bf246ea2c870c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="28bdc41307b4ad9c7a1fa9933d9f6ffb2b9bd66cfeec4a65e6a77f8fb33e7005" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.3.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e189577c21813f797f51c7ff0ea329d448bf3e683b2c16569384f87c6e85c521" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.3.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="19e651cf4b96a637f92231202642cbc268ee45d44011fd1c10fc24c2c89bfaf6" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3c3a90d7ef512cf466edda78054c145f5d672b8d0d4cf1c0a556bb2f26e36763" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2cb9175fba8464914c4caa7bb60f7afc938966eb4fd1389e9049df756f2b3189" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="270304102c2d417b6dceb67bf019d888766cfc51b088ac29beb4ef73c77d6c00" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c8074f5fc3cad33341369215f1afc69a2e35ce60513cbccd18abbe5d60630cf4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="2.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="6f8212ebf50812c475a28e523f11431f8823ea28c33852dcfc035bfef37e0eaa" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="3.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="f84bca54deecb4338a83918c79d742ddbea8de5b928b692f09cf212a8d2c0910" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fa1c908c68fa1a70ea8e02696875843419f60fe9da7a1bb6effab14881887050" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.7.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0d2d95e7776b34eeb4983a68c11a952bbed56afb911895cea9c09112b779298f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896001">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6cc458a4968908eaf124cb859a35016b45d81cac530e44f8af95d83b1edaf4d4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896001">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 19.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/19.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 19.8.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/19.8.0...19.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-19.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="98cdacb6fc3f1068e492d5925e958b790c0d60b006d825aa4f46949db25bb354" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cd7f0cdc0cc16d963cc56ff5ce2a5cbcc311b9ef1f2225ff523ba0e13813892d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c0ebf0848d245d33afe9236837acfe80120f1cf630e876be76e5f3560e134687" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e923e4777facc9f26c26b7a88b5754336aad621aef21ac58ec4e7988bdcb0aa2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="80d587103669ab733376bcef137e01f3b43fbbc4cfbbeb9c7daf8119b2abd1ab" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="22c81fb36639a11ffedef9281b44fb88e86dbcf7f8a45e05c7f82984aa96f70b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="9af5ba54e0ee413efc4ae289437a1bf8199f47e4da8c414195c2254119fc6fb4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="7c63a4d09c6ee86272daf8d835ba09b84e4a63ed5991af67c927d0b1d921679e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="3832cd7853bf1cf1ce443d119292118bf7b66d3ab0d0f94937cbac577ecef16e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="7967e09110f2fead6a2c41a40d461565ef21cd3dd5c036ce3e8f6a7699267649" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="30c6c06db34329d862363c62bd7f989a1d8cadfc66b633aa12de1dd55fba963a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.15.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="8d85a337c226aacdeb84007411a8d7b9cbc666e7ae62686cc93e95d91005ef04" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.15.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="b6bdaeb2a9663bbfe14f1724b27ccb0b999451cad6939008f0df840c2b58d5fd" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.15.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="b6d734dcfb0c6a5317147f9f9895bb16cc9ff5d9f20d477149fc5383855c522d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.16.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="b3379799d7ef037b34866ae305ae069058213a3c6c3470cea4dc445a19e773f6" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.17.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
</package>
<package pkgid="889ece2ad2de9ef955f8b636768c8490c820f809ce27510992a7ee4795c347bc" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.18.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
</package>
<package pkgid="7af070b3523fe0a44abae5a5ba6e6dafe813408bb40371527b5db8172f59f085" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.18.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
</package>
<package pkgid="d03e61752c3e5f16cf25dd0159a941b2457cbdd8217c540c6864499f0e28ee29" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.18.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="cea25b207d7a385497ca319b5b38fc55344f2734aa0da3e9c3ea1a040b1eb3ad" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.19.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="4cb0126012291068da812115a9a4611946ea96e32ea95267f9749882f0667fde" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f96e8722a9cdc33187c1384edde447023ba815e3116d24ef6141b533ba9d5cce" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.2.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="48df6fcd276a057bdc735b1cade01a960a3b9f98ac0e291db92b85d1f72e72f1" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.20.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
</package>
<package pkgid="e5a045a77ab6a044330ebf2b59cd4647531d706f4c61d2e0067254bc38caa22e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.20.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.15.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.2
-
- Change Log for Release asterisk-20.15.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.1...20.15.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.16.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.16.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.16.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.16.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.16.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.16.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.15.2...20.16.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.16.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on
-          u..
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing
-                 URI.
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Upgrade Notes:
- ==============
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.17.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.17.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.17.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.17.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.17.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 34
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
-                  The STIR_SHAKEN_ATTESTATION dialplan function has been added
-                  which will allow suppressing attestation on a call-by-call
-                  basis regardless of the profile attached to the outgoing
-                  endpoint.
-
- func_channel: Allow R/W of ADSI CPE capability setting.
-               CHANNEL(adsicpe) can now be read or written to change the
-               channels' ADSI CPE capability setting.
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- func_math: Add DIGIT_SUM function.
-            The DIGIT_SUM function can be used to return the digit sum of a
-            number.
-
- app_sf: Add post-digit timer option to ReceiveSF.
-         The 't' option for ReceiveSF now allows for a timer since the last
-         digit received, in addition to the number-wide timeout.
-
- app_dial: Allow fractional seconds for dial timeouts.
-           The answer and progress dial timeouts now have millisecond precision,
-           instead of having to be whole numbers.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The threadpool_* options in stasis.conf have now been deprecated
-           though they continue to be read and used. They have been replaced
-           with taskpool options that give greater control over the underlying
-           taskpool used for stasis.
-
- Developer Notes:
- ----------------
-
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
-             A "tech_cause" parameter has been added to the ChannelHangupRequest
-             and ChannelDestroyed ARI event messages and a "TechCause" parameter
-             has been added to the HangupRequest, SoftHangupRequest and Hangup
-             AMI event messages. For chan_pjsip, these will be set to the last
-             SIP response status code for off-nominally terminated calls. The
-             parameter is suppressed for nominal termination.
-
- ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.
-      The ARI /bridges/play and /bridges/record REST APIs have new parameters
-      that allow the caller to specify the format to be used on the "Announcer"
-      and "Recorder" channels respecitvely.
-
- taskpool: Add taskpool API, switch Stasis to using it.
-           The taskpool API has been added for common usage of a pool of
-           taskprocessors. It is suggested to use this API instead of the
-           threadpool+taskprocessor approach.
-
- Commit Authors:
- ---------------
-
- Anthony Minessale: (1)
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.0-1" date="1769083200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0...20.18.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 57
- Commit Authors: 20
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- chan_websocket.conf.sample: Fix category name.
-                             The category name in the chan_websocket.conf.sample
-                             file was incorrect. It should be "global" instead
-                             of "general".
-
- cli.c: Allow 'channel request hangup' to accept patterns.
-        The 'channel request hangup' CLI command now accepts multiple channel
-        names, POSIX Extended Regular Expressions, glob-like patterns, or a
-        combination of all of them. See the CLI command 'core show help channel
-        request hangup' for full details.
-
- res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache
-                           populate command
-                           The AMI command sorcery memory cache populate will
-                           now return an error if there is an internal error
-                           performing the populate. The CLI command will display
-                           an error in this case as well.
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: Two new optional profile parameters have been
-                  added.
-                  * `pidf_element_id` which sets the value of the `id` attribute
-                    on the top-level PIDF-LO `device`, `person` or `tuple`
-                    elements.
-                  * `device_id` which sets the content of the `&lt;deviceID&gt;`
-                    element.
-                  Both parameters can include channel variables.
-
- res_pjsip_messaging: Add support for following 3xx redirects
-                      A new pjsip endpoint option follow_redirect_methods was
-                      added. This option is a comma-delimited, case-insensitive
-                      list of SIP methods for which SIP 3XX redirect responses
-                      are followed. An alembic upgrade script has been added for
-                      adding this new option to the Asterisk database.
-
- taskprocessors: Improve logging and add new cli options
-                 New CLI command has been added - core show taskprocessor name
-                 &lt;taskprocessor-name&gt;
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new "enabled" parameter has been added to ccss.conf. It defaults to
-       "yes" to preserve backwards compatibility but CCSS is rarely used so
-       setting "enabled = no" in the "general" section can save some unneeded
-       channel locking operations and log message spam. Disabling ccss will also
-       prevent the func_callcompletion and chan_dahdi modules from loading.
-
- Makefile: Add module-list-* targets.
-           Try "make module-list-deprecated" to see what modules are on their
-           way out the door.
-
- app_mixmonitor: Add 's' (skip) option to delay recording.
-                 This change introduces a new 's(&lt;seconds&gt;)' (skip) option to
-                 the MixMonitor application. Example:
-                 MixMonitor(${UNIQUEID}.wav,s(3))
-                 This skips recording for the first 3 seconds before writing
-                 audio to the file. Existing MixMonitor behavior remains
-                 unchanged when the 's' option is not used.
-
- app_queue.c: Only announce to head caller if announce_to_first_user
-              When announce_to_first_user is false, no announcements are played
-              to the head caller
-
- Upgrade Notes:
- --------------
-
- res_geolocation: Fix multiple issues with XML generation.
-                  Geolocation: In order to correct bugs in both code and
-                  documentation, the following changes to the parameters for GML
-                  geolocation locations are now in effect:
-                  * The documented but unimplemented `crs` (coordinate reference
-                    system) element has been added to the location_info
-                    parameter that indicates whether the `2d` or `3d` reference
-                    system is to be used. If the crs isn't valid for the shape
-                    specified, an error will be generated. The default depends
-                    on the shape specified.
-                  * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs.
-                    If crs isn't specified, it will default to `2d` for these
-                    shapes. The Sphere, Ellipsoid and Prism shapes MUST use a
-                    `3d` crs. If crs isn't specified, it will default to `3d`
-                    for these shapes. The Point and Polygon shapes may use
-                    either crs. The default crs is `2d` however so if `3d`
-                    positions are used, the crs must be explicitly set to `3d`.
-                  * The `geoloc show gml_shape_defs` CLI command has been
-                    updated to show which coordinate reference systems are valid
-                    for each shape.
-                  * The `pos3d` element has been removed in favor of allowing
-                    the `pos` element to include altitude if the crs is `3d`.
-                    The number of values in the `pos` element MUST be 2 if the
-                    crs is `2d` and 3 if the crs is `3d`. An error will be
-                    generated for any other combination.
-                  * The angle unit-of-measure for shapes that use angles should
-                    now be included in the respective parameter. The default is
-                    `degrees`. There were some inconsistent references to
-                    `orientation_uom` in some documentation but that parameter
-                    never worked and is now removed. See examples below.
-                  Examples...
-                  ```
-                    location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
-                    location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
-                    location_info = shape="Point", pos="39.0 -105.0"
-                    location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
-                                  semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
-                    pidf_element_id = ${CHANNEL(name)}-${EXTEN}
-                    device_id = mac:001122334455
-                    Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
-                  ```
-
- pjsip: Move from threadpool to taskpool
-        The threadpool_* options in pjsip.conf have now been deprecated though
-        they continue to be read and used. They have been replaced with taskpool
-        options that give greater control over the underlying taskpool used for
-        PJSIP. An alembic upgrade script has been added to add these options to
-        realtime as well.
-
- app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE.
-                        In an effort to reduce log spam, two normal progress
-                        "pickup attempted" log messages from app_directed_pickup
-                        have been changed from NOTICE to VERBOSE(3). This puts
-                        them on par with other normal dialplan progress
-                        messages.
-
- Developer Notes:
- ----------------
-
- ccss: Add option to ccss.conf to globally disable it.
-       A new API ast_is_cc_enabled() has been added. It should be used to ensure
-       that CCSS is enabled before making any other ast_cc_* calls.
-
- chan_websocket: Add ability to place a MARK in the media stream.
-                 Apps can now send a `MARK_MEDIA` command with an optional
-                 `correlation_id` parameter to chan_websocket which will be
-                 placed in the media frame queue. When that frame is dequeued
-                 after all intervening media has been played to the core,
-                 chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the
-                 app with the same correlation_id (if any).
-
- chan_websocket: Add capability for JSON control messages and events.
-                 The chan_websocket plain-text control and event messages are
-                 now deprecated (but remain the default) in favor of JSON
-                 formatted messages. See
-                 https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket
-                 for more information. A "transport_data" parameter has been added.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (1)
- C. Maj: (1)
- Daouda Taha: (1)
- Etienne Lessard: (1)
- George Joseph: (12)
- Joe Garlick: (2)
- Joshua C. Colp: (1)
- Justin T. Gibbs: (1)
- Kristian F. Høgh: (1)
- Maximilian Fridrich: (2)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Nathaniel Wesley Filardo: (1)
- Naveen Albert: (3)
- Peter Krall: (1)
- Sean Bright: (17)
- Sven Kube: (1)
- Tinet-mucw: (2)
- phoneben: (5)
- sarangr7: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.1-1" date="1769688000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.18.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.0...20.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
-
- Commit Authors:
- ---------------
-
- Sean Bright: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
-
- 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers
-              when system locale is non-standard
-
- Commits By Author:
- ------------------
-
- Sean Bright (1):
-
- Commit List:
- ------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
-
- Commit Details:
- ---------------
-
- asterisk.c: Use C.UTF-8 locale instead of relying on user's environment.
- Author: Sean Bright
- Date:   2026-01-23
-
- Resolves: #1739</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.18.2-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.18.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.18.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.18.2
-
- Change Log for Release asterisk-20.18.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.18.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.1...20.18.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.18.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-       Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-       Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-       ast_coredumper running as root sources ast_debug_tools.conf from
-       /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-       The Asterisk embedded web server's /httpstatus page echos user supplied
-       values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-       ast_coredumper runs as root, and writes gdb init file to world writeable
-       folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-                 Author: George Joseph
-                 Date:   2026-01-15
-
-                 To address potential security issues, the httpstatus page is
-                 now disabled by default and the echoed query string and cookie
-                 output is html-escaped.
-
-                 Resolves: #GHSA-v6hp-wh3r-cwxh
-
-                 UpgradeNote: To prevent possible security issues, the
-                 `/httpstatus` page served by the internal web server is now
-                 disabled by default. To explicitly enable it, set
-                 `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.19.0-1" date="1775736000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.19.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.19.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.19.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.18.2...20.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 50
- Commit Authors: 21
- Issues Resolved: 34
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.0-1" date="1781179200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.20.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.19.0...20.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 24
- Issues Resolved: 43
- Security Advisories Resolved: 0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.20.1-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 20.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.20.1
-
- Change Log for Release asterisk-20.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.20.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.20.0...20.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 18
- Commit Authors: 6
- Issues Resolved: 0
- Security Advisories Resolved: 19
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-     chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-     Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-     NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-     Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-     Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-     Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-     Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-     Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-     Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-     closes during SDP processing
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-     Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-     Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-     SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-     Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-     in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-     Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count
-     Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-     LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-     Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-     Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-     ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-     ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-     and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-     Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-     2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-     ast_loggrabber executes python script in world writable directory(`/tmp`)
-     leading to potential privilege escalation And RCE
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Mike Bradeen: (2)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Mike Bradeen (2):
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                       address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h</changelog>
</package>
<package pkgid="f6d384427ff05b231331dee24ad8fde558fd88e1610e7425fd81e162a35ce477" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
</package>
<package pkgid="3c585f2513de1741b5d65c3fe6ea937b84fc1441fd7334ae6a08ef05cb1eeca3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="41332c401ff99986fa8a1508f1126015b9b0c86c608a8ab4f17cf15f50666df2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="4c67fd955e46919f34409c8ff7f409883c0ccead232ba6c3adef3a0fa6ebd803" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="a856375d3c231a6893d13a77bb955d9d344ce87e430912789c88aea366724104" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="37f34e59d2b3542a592030f0bbf0723dedb88cef2824021524e2941c83aa1593" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="f2491f6ec92dc0da539aee118e95fadd3c22a3e2432c2d861fd8fb23d2d53035" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="b0edd3caf554e77d0656f07712e3d4010e301350f65dcc2006ffe1d3a749d2a9" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="359a90a418b333c1bb65948bd2d99ee17becfc67e62470b764ea8fb32a91190d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="954abda3a05b79028a771e7f9067e9a3302a69eb5f63f17278601e971166d6e3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="bf1b07398fb0ca5a1d470fc4f0bea038cf50a659c901fee2d64f241277e44ad1" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="f60ed2f4be915316e7ca4744869e7d692f9917bd05588c118c6891bc5804fbe0" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="3627309e52f1367e40f4c651e6520185424ecb90f80f84f9f1f3f2f5cd893583" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="deeb40835f0852f4c6e88904b31e6de7fcee038065fc00f91bc4a06c2fbc5a93" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="667a4c47354fb3acde7b9aa32f944e758a4bf7f22b1c204eb31818e18b2111d4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="3de4d4022794dace97f03081148b0641bb6392a8dee4b25abc244b3e618cb874" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="fa710af1ef64ea0bf9a279f9fab8beeb9448a1bcc8bbb1df8397627328fb705f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="fba18189c2ad9cbd3b1ef493a6e18ee948848353796da15ff389fa5c384c3dec" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="1270af32cabeba0066215b3c21e49edfe21a82add789dd11a2ee7dc7663eafdf" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.10.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1747915201">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="c0e6063e5f32ec7ce42eaf09412ce872aba62a297c6ba76418ebd96fed03abaa" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.10.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
</package>
<package pkgid="f45a660e76fab796d2240841332cd4444546e28e1b407f47cdb916dc596289ea" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.10.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
</package>
<package pkgid="f9ce8d64ae1fe9bd48d974c09b621930d93ecc8e2f30a430d2da3be6109b5bf2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.11.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
</package>
<package pkgid="d8f38873d7518422d6f00ee80b23df1da0c32ee7502ddf66f998984861854ad8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.12.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
</package>
<package pkgid="4cb345b2c2b7558322584f6bbce3f902f5528f291aad874168495ebea39508bb" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.12.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
</package>
<package pkgid="e8a2218f31111646fcb505cbd4761e99da93e7073d32ba20d4969c9c06f38c28" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.12.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
</package>
<package pkgid="2c0ee3b189bd669b0a4b536c88275481d5005a2978d7745848cd3532c1deeba2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.12.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.1
-
- Change Log for Release asterisk-21.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.0...21.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-   [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-      Using malformed From header can forge identity with ";" or NULL in name
-      portion
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-   res_pjsip_messaging.c: Mask control characters in received From display name
-   asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.9.1...21.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-   [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-      cli_permissions.conf: deny option does not work for disallowing shell
-      commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal.
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-    Asterisk can now establish websocket sessions _to_ your ARI applications as
-    well as accepting websocket sessions _from_ them.
-    Full details: http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-    With this release, you can now select an alternate channel storage backend
-    based on C++ Maps. Using the new backend may increase performance and reduce
-    the chances of deadlocks on heavily loaded systems. For more information,
-    see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.1-1" date="1753963200">- The Asterisk Development Team would like to announce security release Asterisk
- 22.5.1.
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.5.1
- https://github.com/asterisk/asterisk/releases/tag/21.10.1
- https://github.com/asterisk/asterisk/releases/tag/20.15.1
- https://github.com/asterisk/asterisk/releases/tag/18.26.3
- https://github.com/asterisk/asterisk/releases/tag/certified-20.7-cert7
-
- and
-
- https://downloads.asterisk.org/pub/telephony/asterisk
- Repository: https://github.com/asterisk/asterisk
- Tags: 22.5.1, 21.10.1, 20.15.1, 18.26.3, certified-20.7-cert7
-
- Change Log
- ==========
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.5.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.0...22.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
- [Downloads] (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.5.1.tar.gz)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
-    GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-    asterisk/res/res_stir_shaken/verification.c
-    GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk
-    script may allow local privilege escalation.
-
- Upgrade Notes:
- --------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-                The safe_asterisk script now checks that, if it was run by the
-                root user, the /etc/asterisk/startup.d directory and all the
-                files it contains are owned by root. If the checks fail,
-                safe_asterisk will exit with an error and Asterisk will not be
-                started. Additionally, the default logging destination is now
-                stderr instead of tty "9" which probably won't exist in modern
-                systems.

- Commit Authors:
- ---------------
-
- George Joseph: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in
-                       asterisk/res/res_stir_shaken/verification.c
- !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script
-                       may allow local privilege escalation.
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_stir_shaken: Test for missing semicolon in Identity header.
- ThatTotallyRealMyth (1):
-    safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its
-    files.
-
- Commit List:
- ------------
-
- safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
- res_stir_shaken: Test for missing semicolon in Identity header.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.10.2-1" date="1756382400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.10.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.10.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.10.2
-
- Change Log for Release asterisk-21.10.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.10.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.1...21.10.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.10.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-    [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j):
-    A specifically malformed Authorization header in an incoming SIP request can
-    cause Asterisk to crash
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-64qc-9x89-rx5j: A specifically malformed Authorization header in an
-                       incoming SIP request can cause Asterisk to crash
-
- Commits By Author:
- ------------------
-
- George Joseph (1):
-    res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns
-    NULL.
-
- Commit List:
- ------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-
- Commit Details:
- ---------------
-
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
-                                 Author: George Joseph
-                                 Date:   2025-08-28
-
-                                 In the highly-unlikely event that
-                                 get_authorization_hdr() couldn't find an
-                                 Authorization header in a request, trying to
-                                 get the digest algorithm would cauase a SEGV.
-                                 We now check that we have an auth header that
-                                 matches the realm before trying to get the
-                                 algorithm from it.
-
-                                 Resolves: #GHSA-64qc-9x89-rx5j</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.11.0-1" date="1760529600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.11.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.10.2...21.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 54
- Commit Authors: 22
- Issues Resolved: 40
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- app_queue.c: Add new global 'log_unpause_on_reason_change'
-              Add new global option 'log_unpause_on_reason_change' that is
-              default disabled. When enabled cause addition of UNPAUSE event on
-              every re-PAUSE with reason changed.
-
- pbx_builtins: Allow custom tone for WaitExten.
-               The tone used while waiting for digits in WaitExten can now be
-               overridden by specifying an argument for the 'd' option.
-
- res_tonedetect: Add option for TONE_DETECT detection to auto stop.
-                 The 'e' option for TONE_DETECT now allows detection to be
-                 disabled automatically once the desired number of matches have
-                 been fulfilled, which can help prevent race conditions in the
-                 dialplan, since TONE_DETECT does not need to be disabled after
-                 a hit.
-
- sorcery: Prevent duplicate objects and ensure missing objects are created on u.
-          Users relying on Sorcery multiple writable backends configurations
-          (e.g., astdb + realtime) may now enable
-          update_or_create_on_update_miss = yes in sorcery.conf to ensure
-          missing objects are recreated after temporary backend failures.
-          Default behavior remains unchanged unless explicitly enabled.
-
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI
-                 A new WebSocket channel driver option `v` has been added to the
-                 Dial application that allows you to specify additional URI
-                 parameters on outgoing connections. Run `core show application
-                 Dial` from the Asterisk CLI to see how to use it.
-
- app_chanspy: Add option to not automatically answer channel.
-              ChanSpy and ExtenSpy can now be configured to not automatically
-              answer the channel by using the 'N' option.
-
- cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
-      Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types
-      in cel.conf will log media files and music on hold played to each channel.
-      The STREAM_BEGIN event's extra field will contain a JSON with the file
-      details (path, format and language), or the class name, in case of music
-      on hold is played. The DTMF event's extra field will contain a JSON with
-      the digit and the duration in milliseconds.
-
- res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM
-           Options are now available in the menuselect "Resource Modules"
-           category that allow you to enable the AES_192, AES_256 and AES_GCM
-           cipher suites in res_srtp. Of course, libsrtp and OpenSSL must
-           support them but modern versions do. Previously, the only way to
-           enable them was to set the CFLAGS environment variable when running
-           ./configure. The default setting is to disable them preserving
-           existing behavior.
-
- cdr: add CANCEL dispostion in CDR
-      A new CDR option "canceldispositionenabled" has been added that when set
-      to true, the NO ANSWER disposition will be split into two dispositions:
-      CANCEL and NO ANSWER. The default value is 'no'
-
- func_curl: Allow auth methods to be set.
-            The httpauth field in CURLOPT now allows the authentication methods
-            to be set.
-
- Media over Websocket Channel Driver
-    A new channel driver "chan_websocket" is now available. It can exchange
-    media over both inbound and outbound websockets and will both frame and
-    re-time the media it receives.
-    See http://s.asterisk.net/mow for more information.
-    The ARI channels/externalMedia API now includes support for the
-
- Developer Notes:
- ----------------
-
- ARI: Add command to indicate progress to a channel
-      A new ARI endpoint is available at `/channels/{channelId}/progress` to
-      indicate progress to a channel.
-
- options: Change ast_options from ast_flags to ast_flags64.
-          The 32-bit ast_options has no room left to accomodate new options and
-          so has been converted to an ast_flags64 structure. All internal
-          references to ast_options have been updated to use the 64-bit flag
-          manipulation macros. External module references to the 32-bit
-          ast_options should continue to work on little-endian systems because
-          the least-significant bytes of a 64 bit integer will be in the same
-          location as a 32-bit integer. Because that's not the case on
-          big-endian systems, we've swapped the bytes in the flags manupulation
-          macros on big-endian systems so external modules should still work
-          however you are encouraged to test.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Alexey Khabulyak: (2)
- Allan Nathanson: (1)
- Artem Umerov: (1)
- Ben Ford: (1)
- George Joseph: (12)
- Igor Goncharovsky: (2)
- Jaco Kroon: (1)
- Joe Garlick: (1)
- Jose Lopes: (1)
- Kodokaii: (1)
- Martin Tomec: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (15)
- Sean Bright: (2)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (1)
- Stuart Henderson: (1)
- Sven Kube: (2)
- Tinet-Mucw: (2)
- Zhou_jiajian: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.0-1" date="1763640000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 10
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
-                     Added a new option to HANGUPCAUSE to access additional
-                     information about hangup reason. Reason headers from pjsip
-                     could be read using 'tech_extended' cause type.
-
- chan_dahdi: Add DAHDI_CHANNEL function.
-             The DAHDI_CHANNEL function allows for getting/setting certain
-             properties about DAHDI channels from the dialplan.
-
- Upgrade Notes:
- --------------
-
- res_audiosocket: add message types for all slin sample rates
-                  New audiosocket message types 0x11 - 0x18 has been added for
-                  slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
-                  slin192 audio. External applications using audiosocket may
-                  need to be updated to support these message types if the
-                  audiosocket channel is created with one of these audio
-                  formats.
-
- Commit Authors:
- ---------------
-
- Bastian Triller: (1)
- Ben Ford: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (4)
- Sean Bright: (3)
- Sven Kube: (3)
- phoneben: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.1-1" date="1770292800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.1
-
- Change Log for Release asterisk-21.12.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.0...21.12.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 4
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 4
-    [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42):
-      Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE
-      Injection
-    [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3):
-      ast_coredumper running as root sources ast_debug_tools.conf from
-      /etc/asterisk; potentially leading to privilege escalation
-    [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh):
-      The Asterisk embedded web server's /httpstatus page echos user supplied
-      values(cookie and query string) without sanitization
-    [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c):
-      ast_coredumper runs as root, and writes gdb init file to world writeable
-      folder; leading to potential privilege escalation
-
- User Notes:
- -----------
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 ast_debug_tools.conf must be owned by root and not be writable
-                 by other users or groups to be used by ast_coredumper or by
-                 ast_logescalator or ast_loggrabber when run as root.
-
- Upgrade Notes:
- --------------
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         To prevent possible security issues, the `/httpstatus` page served by
-         the internal web server is now disabled by default. To explicitly
-         enable it, set `enable_status=yes` in http.conf.
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
- Mike Bradeen: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-85x7-54wr-vh42: Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to
-                       potential XXE Injection
- !GHSA-rvch-3jmx-3jf3: ast_coredumper running as root sources
-                       ast_debug_tools.conf from /etc/asterisk; potentially
-                       leading to privilege escalation
- !GHSA-v6hp-wh3r-cwxh: The Asterisk embedded web server's /httpstatus page echos
-                       user supplied values(cookie and query string) without
-                       sanitization
- !GHSA-xpc6-x892-v83c: ast_coredumper runs as root, and writes gdb init file to
-                       world writeable folder; leading to potential privilege
-                       escalation
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
- Mike Bradeen (2):
-
- Commit List:
- ------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
- ast_coredumper: check ast_debug_tools.conf permissions
- http.c: Change httpstatus to default disabled and sanitize output.
- ast_coredumper: create gdbinit file with restrictive permissions
-
- Commit Details:
- ---------------
-
- xml.c: Replace XML_PARSE_NOENT with XML_PARSE_NONET for xmlReadFile.
-        Author: George Joseph
-        Date:   2026-01-15
-
-        The xmlReadFile XML_PARSE_NOENT flag, which allows parsing of external
-        entities, could allow a potential XXE injection attack. Replacing it
-        with XML_PARSE_NONET, which prevents network access, is safer.
-
-        Resolves: #GHSA-85x7-54wr-vh42
-
- ast_coredumper: check ast_debug_tools.conf permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Prevent ast_coredumper from using ast_debug_tools.conf files
-                 that are not owned by root or are writable by other users or
-                 groups.
-
-                 Prevent ast_logescalator and ast_loggrabber from doing the same
-                 if they are run as root.
-
-                 Resolves: #GHSA-rvch-3jmx-3jf3
-
-                 UserNote: ast_debug_tools.conf must be owned by root and not be
-                 writable by other users or groups to be used by ast_coredumper
-                 or by ast_logescalator or ast_loggrabber when run as root.
-
- http.c: Change httpstatus to default disabled and sanitize output.
-         Author: George Joseph
-         Date:   2026-01-15
-
-         To address potential security issues, the httpstatus page is now
-         disabled by default and the echoed query string and cookie output is
-         html-escaped.
-
-         Resolves: #GHSA-v6hp-wh3r-cwxh
-
-         UpgradeNote: To prevent possible security issues, the `/httpstatus`
-         page served by the internal web server is now disabled by default. To
-         explicitly enable it, set `enable_status=yes` in http.conf.
-
- ast_coredumper: create gdbinit file with restrictive permissions
-                 Author: Mike Bradeen
-                 Date:   2026-01-15
-
-                 Modify gdbinit to use the install command with explicit
-                 permissions (-m 600) when creating the .ast_coredumper.gdbinit
-                 file. This ensures the file is created with restricted
-                 permissions (readable/writable only by the owner) to avoid
-                 potential privilege escalation.
-
-                 Resolves: #GHSA-xpc6-x892-v83c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.2-1" date="1774526400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.12.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.2
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.12.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.1...21.12.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- Mike Bradeen: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 1833: [bug]: Address security vulnerabilities in pjproject
-
- Commits By Author:
- ------------------
-
- Mike Bradeen (1):
-  - res_pjsip: Address pjproject security vulnerabilities
-
- Commit List:
- ------------
-
- res_pjsip: Address pjproject security vulnerabilities
-
- Commit Details:
- ---------------
-
- res_pjsip: Address pjproject security vulnerabilities
-            Author: Mike Bradeen
-            Date:   2026-03-25
-
-            Address the following pjproject security vulnerabilities
-
-            [GHSA-j29p-pvh2-pvqp - Buffer overflow in ICE with long username](https://github.com/pjsip/pjproject/security/advisories/GHSA-j29p-pvh2-pvqp)
-            [GHSA-8fj4-fv9f-hjpc - Heap use-after-free in PJSIP presense subscription termination header](https://github.com/pjsip/pjproject/security/advisories/GHSA-8fj4-fv9f-hjpc)
-            [GHSA-g88q-c2hm-q7p7 - ICE session use-after-free race conditions](https://github.com/pjsip/pjproject/security/advisories/GHSA-g88q-c2hm-q7p7)
-            [GHSA-x5pq-qrp4-fmrj - Out-of-bounds read in SIP multipart parsing](https://github.com/pjsip/pjproject/security/advisories/GHSA-x5pq-qrp4-fmrj)
-
-            Resolves: #1833</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.12.3-1" date="1782388800">- The Asterisk Development Team would like to announce security release Asterisk
- 21.12.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.12.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.12.3
-
-  Change Log for Release asterisk-21.12.3
- ========================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.3.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.12.2...21.12.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 21
- Commit Authors: 7
- Issues Resolved: 0
- Security Advisories Resolved: 20
-  [GHSA-3g56-cgrh-95p5](https://github.com/asterisk/asterisk/security/advisories/GHSA-3g56-cgrh-95p5):
-   chan_unistim DIALPAGE digit handling can overflow phone_number and crash
-   Asterisk
-  [GHSA-3rhj-hhw7-m6fw](https://github.com/asterisk/asterisk/security/advisories/GHSA-3rhj-hhw7-m6fw):
-   NULL Pointer Dereference in HTTP AMI Digest Authentication
-  [GHSA-4pgv-j3mr-3rcp](https://github.com/asterisk/asterisk/security/advisories/GHSA-4pgv-j3mr-3rcp):
-   Reflected XSS in Phone Provisioning HTTP Error Pages
-  [GHSA-589g-qgf8-m6mx](https://github.com/asterisk/asterisk/security/advisories/GHSA-589g-qgf8-m6mx):
-   Stack buffer overflow in MWI NOTIFY Message-Account parsing
-  [GHSA-746q-794h-cc7f](https://github.com/asterisk/asterisk/security/advisories/GHSA-746q-794h-cc7f):
-   Out-of-Bounds Read in Q.931 Information Element Parser (H.323 Addon)
-  [GHSA-8jhw-m2hg-vp3h](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jhw-m2hg-vp3h):
-   Heap Buffer Overflow in OGG/Speex File Playback (format_ogg_speex)
-  [GHSA-8jw3-ccr9-xrmf](https://github.com/asterisk/asterisk/security/advisories/GHSA-8jw3-ccr9-xrmf):
-   Buffer over-read in Asterisk PJSIP MWI body parser
-  [GHSA-g8q2-p36q-94f6](https://github.com/asterisk/asterisk/security/advisories/GHSA-g8q2-p36q-94f6):
-   Heap-use-after-free in Asterisk PJSIP TCP/SDP handling when TCP connection
-   closes during SDP processing
-  [GHSA-h5hv-jmgj-92q2](https://github.com/asterisk/asterisk/security/advisories/GHSA-h5hv-jmgj-92q2):
-   CVE-2022-37325 fix is absent from current chan_ooh323 Q.931 party-number
-   parser
-  [GHSA-j2mm-57pq-jh94](https://github.com/asterisk/asterisk/security/advisories/GHSA-j2mm-57pq-jh94):
-   Possible RED T.140 Generation Accumulation OOB Write
-  [GHSA-mxgm-8c6f-5p8f](https://github.com/asterisk/asterisk/security/advisories/GHSA-mxgm-8c6f-5p8f):
-   Stack buffer overflow in res_xmpp XMPP namespace prefix handling
-  [GHSA-ph27-3m5q-mj5m](https://github.com/asterisk/asterisk/security/advisories/GHSA-ph27-3m5q-mj5m):
-   SQL Injection in cel_pgsql and cel_tds via CELGenUserEvent eventtype Field
-  [GHSA-q9fr-m7g8-6ph5](https://github.com/asterisk/asterisk/security/advisories/GHSA-q9fr-m7g8-6ph5):
-   Asterisk app_sms.c copies externally controlled SMS lengths into fixed
-   in-struct buffers
-  [GHSA-qf8j-jp7h-c5hx](https://github.com/asterisk/asterisk/security/advisories/GHSA-qf8j-jp7h-c5hx):
-   Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil Sample Count Mismatch
-  [GHSA-r6c2-hwc2-j4mp](https://github.com/asterisk/asterisk/security/advisories/GHSA-r6c2-hwc2-j4mp):
-   LDAP Filter Injection in res_config_ldap via SIP Username (Unauthenticated
-   Information Disclosure)
-  [GHSA-vfhr-r9x9-c687](https://github.com/asterisk/asterisk/security/advisories/GHSA-vfhr-r9x9-c687):
-   Possible RED T.140 Heap Buffer Overflow
-  [GHSA-vrfp-mg3q-3959](https://github.com/asterisk/asterisk/security/advisories/GHSA-vrfp-mg3q-3959):
-   ARI setChannelVar bypasses live_dangerously and permits FILE() writes
-  [GHSA-wcvv-g26m-wx5c](https://github.com/asterisk/asterisk/security/advisories/GHSA-wcvv-g26m-wx5c):
-   ARI REST-over-WebSocket read-only bypass allows arbitrary module path load
-   and conditional RCE
-  [GHSA-x348-j6c9-77f3](https://github.com/asterisk/asterisk/security/advisories/GHSA-x348-j6c9-77f3):
-   Stack Buffer Overflow in H.323 ooTrace() via Unbounded vsprintf into Fixed
-   2048-byte Buffer
-  [GHSA-xgj6-2gc5-5x9c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xgj6-2gc5-5x9c):
-   ast_loggrabber executes python script in world writable directory(`/tmp`)
-   leading to potential privilege escalation And RCE
-
- User Notes:
- -----------
-
- acl: Add ACL support to http and ari
-      A new section, type=restriction has been added to http.conf to allow an
-      uri prefix based acl to be configured. See http.conf.sample for examples
-      and more information. The user section of ari.conf can now contain an acl
-      configuration to restrict users access. See ari.conf.sample for examples
-      and more information
-
- Developer Notes:
- ----------------
-
- ARI: Make ARI applications respect live_dangerously.
-      ARI applications can no longer call "dangerous" dialplan functions like
-      DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling
-      "live_dangerously" in asterisk.conf.
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- Commit Authors:
- ---------------
-
- George Joseph: (6)
- Joshua C. Colp: (1)
- Mike Bradeen: (4)
- Milan Kyselica: (7)
- Pengpeng Hou: (1)
- Roberto Paleari: (1)
- ThatTotallyRealMyth: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-3g56-cgrh-95p5: chan_unistim DIALPAGE digit handling can overflow
-                       phone_number and crash Asterisk
- !GHSA-3rhj-hhw7-m6fw: NULL Pointer Dereference in HTTP AMI Digest
-                       Authentication
- !GHSA-4pgv-j3mr-3rcp: Reflected XSS in Phone Provisioning HTTP Error Pages
- !GHSA-589g-qgf8-m6mx: Stack buffer overflow in MWI NOTIFY Message-Account
-                       parsing
- !GHSA-746q-794h-cc7f: Out-of-Bounds Read in Q.931 Information Element Parser
-                       (H.323 Addon)
- !GHSA-8jhw-m2hg-vp3h: Heap Buffer Overflow in OGG/Speex File Playback
-                       (format_ogg_speex)
- !GHSA-8jw3-ccr9-xrmf: Buffer over-read in Asterisk PJSIP MWI body parser
- !GHSA-g8q2-p36q-94f6: Heap-use-after-free in Asterisk PJSIP TCP/SDP handling
-                       when TCP connection closes during SDP processing
- !GHSA-h5hv-jmgj-92q2: CVE-2022-37325 fix is absent from current chan_ooh323
-                       Q.931 party-number parser
- !GHSA-j2mm-57pq-jh94: Possible RED T.140 Generation Accumulation OOB Write
- !GHSA-mxgm-8c6f-5p8f: Stack buffer overflow in res_xmpp XMPP namespace prefix
-                       handling
- !GHSA-ph27-3m5q-mj5m: SQL Injection in cel_pgsql and cel_tds via
-                       CELGenUserEvent eventtype Field
- !GHSA-q9fr-m7g8-6ph5: Asterisk app_sms.c copies externally controlled SMS
-                       lengths into fixed in-struct buffers
- !GHSA-qf8j-jp7h-c5hx: Out-of-Bounds Write in Codec2 Decoder Due to Floor/Ceil
-                       Sample Count Mismatch
- !GHSA-r6c2-hwc2-j4mp: LDAP Filter Injection in res_config_ldap via SIP Username
-                       (Unauthenticated Information Disclosure)
- !GHSA-vfhr-r9x9-c687: Possible RED T.140 Heap Buffer Overflow
- !GHSA-vrfp-mg3q-3959: ARI setChannelVar bypasses live_dangerously and permits
-                       FILE() writes
- !GHSA-wcvv-g26m-wx5c: ARI REST-over-WebSocket read-only bypass allows arbitrary
-                       module path load and conditional RCE
- !GHSA-x348-j6c9-77f3: Stack Buffer Overflow in H.323 ooTrace() via Unbounded
-                       vsprintf into Fixed 2048-byte Buffer
- !GHSA-xgj6-2gc5-5x9c: ast_loggrabber executes python script in world writable
-                       directory(`/tmp`) leading to potential privilege
-                       escalation And RCE
-
- Commits By Author:
- ------------------
-
- George Joseph (6):
-  chan_unistim.c: Prevent overrun of phone_number field.
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-
- Joshua C. Colp (1):
-  build: Fix GCC discarded-qualifiers const errors.
-
- Mike Bradeen (4):
-  ooh323c: not checking for IE minimum length
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  acl: Add ACL support to http and ari
-
- Milan Kyselica (7):
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-
- Pengpeng Hou (1):
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-
- Roberto Paleari (1):
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-
- ThatTotallyRealMyth (1):
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-
- Commit List:
- ------------
-
-  ast_loggrabber: Install the ast_tsconvert.py script to a secure temp
-                  directory.
-  chan_unistim.c: Prevent overrun of phone_number field.
-  ooh323c: not checking for IE minimum length
-  res_ari: Ensure read-only users are properly authorized via REST Over
-           WebSocket.
-  pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local
-                        address.
-  ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-  ARI: Make ARI applications respect live_dangerously.
-  res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-  res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-  manager: Use remote address in user error logging
-  ooh323: Prevent potential buffer overflow in trace logging
-  app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-  res_xmpp: Fix stack buffer overflow in namespace prefix handling
-  res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-  res_config_ldap: Escape LDAP filter values per RFC 4515
-  cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-  http: Escape error page text to prevent reflected XSS
-  codec_codec2: Only process complete Codec2 frames in decoder
-  format_ogg_speex: Add bounds check to prevent heap buffer overflow
-  acl: Add ACL support to http and ari
-  build: Fix GCC discarded-qualifiers const errors.
-
- Commit Details:
- ---------------
-
- ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory.
-                 Author: ThatTotallyRealMyth
-                 Date:   2026-03-19
-
-                 The ast_tsconvert.py script called by ast_loggrabber is now
-                 installed in a temporary directory that isn't world readable or
-                 writable.
-
-                 Resolves: #GHSA-xgj6-2gc5-5x9c
-
- chan_unistim.c: Prevent overrun of phone_number field.
-                 Author: George Joseph
-                 Date:   2026-06-15
-
-                 Add a check to key_dial_page() to ensure that dialed digits
-                 won't overrun the phone_number field.
-
-                 Resolves: #GHSA-3g56-cgrh-95p5
-
- ooh323c: not checking for IE minimum length
-          Author: Mike Bradeen
-          Date:   2022-06-06
-
-          When decoding q.931 encoded calling/called number now checking for
-          length being less than minimum required.
-
-          Resolves: #GHSA-h5hv-jmgj-92q2
-
- res_ari: Ensure read-only users are properly authorized via REST Over
-          WebSocket.
-          Author: George Joseph
-          Date:   2026-06-12
-
-          The REST over WebSocket path now properly prevents non-GET methods
-          from being executed on inbound WebSockets.
-
-          * The query parameters from the original incoming GET request that
-            caused the upgrade to WebSocket are now passed to all REST requests
-            that come from the client. This ensures that if the client
-            authenticated with a read-only userid using the "api_key"
-            query_string parameter, REST requests coming in over the WebSocket
-            will only be able to execute GETs on resources. The HTTP headers
-            were already passed to the REST requests so if the client had
-            authenticated via an "Authorization" it was properly handled.
-
-          * New tests have been added to test_ari.c to check that read-only
-            users are properly denied access to resources using non-GET methods.
-            Several memory leaks were also squashed.
-
-          Resolves: #GHSA-wcvv-g26m-wx5c
-
- pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address.
-                       Author: George Joseph
-                       Date:   2026-06-10
-
-                       The filter_on_tx_message() function was using
-                       pj_strassign() to save the pointer of the pjproject
-                       transport local address to a local pj_str_t variable.
-                       That variable was ultimately used to set the Contact
-                       header's uri-&gt;host and the SDP connection attribute's
-                       address again using pj_strassign. pj_strassign() doesn't
-                       copy the actual value of the pj_str_t however, it just
-                       copies the pointer so if a connection-oriented transport
-                       is disconnected before the 200 OK with the SDP is sent,
-                       those pointers will be invalid which can cause
-                       use-after-free issues. To prevent this,
-                       filter_on_tx_message() now uses pj_strdup with the
-                       tdata-&gt;pool as the backing store to save the local IP
-                       address to the local variable. pj_strassign() can then be
-                       used safely later on since the tdata will be available
-                       for the life of the transaction.
-
-                       Resolves: #GHSA-g8q2-p36q-94f6
-
- ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds.
-                   Author: George Joseph
-                   Date:   2026-06-02
-
-                   Several bounds checks have been edded to ooQ931Decode to
-                   prevent it from running past the end of the data buffer when
-                   parsing information elements.
-
-                   Resolves: #GHSA-746q-794h-cc7f
-
- ARI: Make ARI applications respect live_dangerously.
-      Author: George Joseph
-      Date:   2026-05-21
-
-      DeveloperNote: ARI applications can no longer call "dangerous" dialplan
-      functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without
-      enabling "live_dangerously" in asterisk.conf.
-
-      Resolves: #GHSA-vrfp-mg3q-3959
-
- res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns.
-                     Author: George Joseph
-                     Date:   2026-04-27
-
-                     * Add check to red_t140_to_red() to ensure that the new
-                       primary payload can't cause the rtp_red-&gt;len array items
-                       to wrap or cause an overrun of the rtp_red-&gt;t140red_data
-                       buffer.
-
-                     * Add check to rtp_red_buffer() to ensure that a T.140
-                       frame to be sent can't cause rtp_red-&gt;len array items to
-                       wrap or cause an overrun of the rtp_red-&gt;buf_data buffer.
-
-                     Resolves: #GHSA-vfhr-r9x9-c687
-                     Resolves: #GHSA-j2mm-57pq-jh94
-
- res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser
-                         Author: Roberto Paleari
-                         Date:   2026-04-29
-
-                         Add constraint checks to prevent unauthenticated users
-                         from crashing Asterisk instance by sending a crafted
-                         inbound SIP NOTIFY request with "Content-Type:
-                         application/simple-message-summary".
-
-                         Resolves: #GHSA-8jw3-ccr9-xrmf
-
- manager: Use remote address in user error logging
-          Author: Mike Bradeen
-          Date:   2026-03-30
-
-          To avoid a potential null dereference use the remote address in error
-          logging when there is no user or the user acl fails.
-
-          Resolves: #GHSA-3rhj-hhw7-m6fw
-
- ooh323: Prevent potential buffer overflow in trace logging
-         Author: Mike Bradeen
-         Date:   2026-03-31
-
-         Replace a call to vsprintf with a call to ast_vasprintf to prevent a
-         possible buffer overflow.
-
-         Resolves: #GHSA-x348-j6c9-77f3
-
- app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers
-          Author: Pengpeng Hou
-          Date:   2026-04-01
-
-          The protocol 1 unpack helpers trusted externally controlled lengths
-          and wrote them directly into fixed-size buffers in sms_t. Clamp the
-          address, header, and body copies to the destination array sizes so
-          malformed messages cannot overwrite adjacent state.
-
-          Resolves: #GHSA-q9fr-m7g8-6ph5
-
- res_xmpp: Fix stack buffer overflow in namespace prefix handling
-           Author: Milan Kyselica
-           Date:   2026-03-26
-
-           The snprintf size parameter in xmpp_action_hook() is computed from
-           the attacker-controlled namespace prefix length and is not bounded by
-           the 256-byte stack buffer size. When a remote XMPP peer sends a
-           stanza with a child element whose namespace prefix exceeds 249
-           characters, snprintf writes past the buffer boundary.
-
-           Use sizeof(attr) as the snprintf size limit and %.*s precision to
-           extract only the prefix portion of the element name, preserving the
-           original truncation behavior for valid inputs.
-
-           Resolves: #GHSA-mxgm-8c6f-5p8f
-
- res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser
-                   Author: Milan Kyselica
-                   Date:   2026-03-24
-
-                   The parse_simple_message_summary() function uses sscanf with
-                   an unbounded %s format specifier to parse the Message-Account
-                   field from incoming SIP NOTIFY bodies into a fixed-size
-                   512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single
-                   unauthenticated SIP NOTIFY with a Message-Account value
-                   exceeding 512 bytes overflows the buffer, corrupting adjacent
-                   stack data and permanently disabling the PJSIP transport
-                   layer without crashing the process.
-
-                   Add a width specifier (%511s) to limit the sscanf write to
-                   PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator,
-                   matching the destination buffer size.
-
-                   Resolves: #GHSA-589g-qgf8-m6mx
-
- res_config_ldap: Escape LDAP filter values per RFC 4515
-                  Author: Milan Kyselica
-                  Date:   2026-03-23
-
-                  The LDAP realtime driver constructs search filters by directly
-                  concatenating user-supplied values without RFC 4515 escaping.
-                  When LDAP is used as a realtime backend for endpoint
-                  identification, characters with special meaning in LDAP
-                  filters (*, (, ), \) can be injected via the SIP From header
-                  username.
-
-                  Add ldap_filter_escape_value() that escapes RFC 4515 special
-                  characters to their \HH hex representation, and apply it to
-                  non-LIKE query values. The LIKE query path preserves the
-                  existing wildcard conversion behavior with a note for
-                  maintainers.
-
-                  Resolves: #GHSA-r6c2-hwc2-j4mp
-
- cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection
-                     Author: Milan Kyselica
-                     Date:   2026-03-23
-
-                     The eventtype column handler in cel_pgsql.c inserts
-                     record.user_defined_name directly into the SQL query
-                     without calling PQescapeStringConn(), while all other
-                     string fields in the same function are properly escaped.
-                     Similarly, cel_tds.c passes the raw user_defined_name into
-                     the SQL INSERT without routing it through anti_injection(),
-                     while all other fields are processed through that function.
-
-                     For cel_pgsql.c, escape the eventtype value using
-                     PQescapeStringConn(), matching the existing pattern used
-                     for all other string fields at lines 308-331 of the same
-                     function.
-
-                     For cel_tds.c, route the eventtype value through
-                     anti_injection() consistent with how all other fields are
-                     handled in the same function.
-
-                     Resolves: #GHSA-ph27-3m5q-mj5m
-
- http: Escape error page text to prevent reflected XSS
-       Author: Milan Kyselica
-       Date:   2026-04-08
-
-       The text parameter in ast_http_create_response() is inserted into the
-       HTML body without escaping, while the server name on the same page is
-       properly escaped via ast_xml_escape(). When res_phoneprov passes the
-       decoded request URI as the text of a 404 response, HTML metacharacters in
-       the URI are rendered by the browser.
-
-       Apply ast_xml_escape() to the text parameter before inserting it into the
-       HTML template, using the same function already used for the server name.
-
-       Resolves: #GHSA-4pgv-j3mr-3rcp
-
- codec_codec2: Only process complete Codec2 frames in decoder
-               Author: Milan Kyselica
-               Date:   2026-04-08
-
-               The codec2_samples() function uses floor division (160 *
-               datalen/6) to compute expected output samples, but the decode
-               loop condition (x &lt; datalen) iterates with ceiling behavior when
-               datalen is not a multiple of CODEC2_FRAME_LEN. This mismatch
-               causes the loop to decode one extra frame beyond what the
-               framework bounds check budgeted for, leading to an out-of-bounds
-               write on the output buffer.
-
-               Change the loop condition to only process complete frames,
-               matching the floor-division behavior of codec2_samples(). This
-               also prevents an out-of-bounds read on the input side when fewer
-               than CODEC2_FRAME_LEN bytes remain.
-
-               Resolves: #GHSA-qf8j-jp7h-c5hx
-
- format_ogg_speex: Add bounds check to prevent heap buffer overflow
-                   Author: Milan Kyselica
-                   Date:   2026-03-23
-
-                   The ogg_speex_read() function copies OGG packet data via
-                   memcpy() without validating the packet size against the
-                   destination buffer (BUF_SIZE = 200 bytes). A crafted .spx
-                   file with an oversized OGG audio packet causes a heap buffer
-                   overflow that corrupts the adjacent speex_desc structure
-                   containing libogg heap pointers, leading to a crash (SIGSEGV)
-                   on playback.
-
-                   Add a bounds check for both negative and oversized values
-                   before the memcpy, consistent with how format_ogg_vorbis
-                   bounds its reads via ov_read().
-
-                   Resolves: #GHSA-8jhw-m2hg-vp3h
-
- acl: Add ACL support to http and ari
-      Author: Mike Bradeen
-      Date:   2026-02-27
-
-      Add uri prefix based acl support to the built in http server. This allows
-      an acl to be added per uri prefix (ie '/metrics' or '/ws') to restrict
-      access.
-
-      Add user based acl support for ARI. This adds new acl options to the user
-      section of ari.conf to restrict access on a per user basis.
-
-      resolves: #1799
-
-      UserNote: A new section, type=restriction has been added to http.conf to
-      allow an uri prefix based acl to be configured. See http.conf.sample for
-      examples and more information. The user section of ari.conf can now
-      contain an acl configuration to restrict users access. See ari.conf.sample
-      for examples and more information
-
- build: Fix GCC discarded-qualifiers const errors.
-        Author: Joshua C. Colp
-        Date:   2026-02-12
-
-        GCC 15.2.1 pays attention to the discarding of the const qualifier when
-        strchr, strrchr, memchr, or memrchr are now used. This change fixes
-        numerous errors with this throughout the tree. The fixes can be broken
-        down into the following:
-
-        1. The return value should be considered const.
-        2. The value passed to strchr or strrchr can be cast as it is expected
-           and allowed to be modified.
-        3. The pointer passed to strchr or strrchr is not meant to be modified
-           and so the contents must be duplicated.
-        4. It was declared const and never should have been.</changelog>
</package>
<package pkgid="fac73599b88b0cb119fd73eb171d889ab3cd857aafaf8e168d2be8ea2360f9a5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="ac726911b97e19de2a1c84cbb1f600671cd9ab11cb6134d1490e147768f8066d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="f161fb2144b65b7f1e1bab3577244b1ee983539f109c3e08a7f0280a991db65d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="0f17c9bc34d2a317b6a5332e8ddad4c0ec4a44bc3c51ecc2f57ce8664217c628" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="4c1c9520cc03e2a918d9658a67b75f1ca161af071c882c6fdf030a3f6f8d49d0" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="54692b9b4e323377f95153b0f76ef12486127f58c593d017ab7b99301fd37c11" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="88fbfd51b030ac7966b114eaf936cfa071f41f0130c42097d091a897538a2cff" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="76bf0e4475f86fad796ea486793d2e8bd8f7dfe9eaf9514fca87dbe9921e4582" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="e0db75d4cf87c1b672d13402c849948d8ee793b41f959f067835b3011a21d6e5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el8"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: ma